search for: ast_debug

...13 20:25:27.233000000 +0200 @@ -407,6 +407,7 @@ /* We have a BYE or CANCEL request without a serializer. */ pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), rdata, PJSIP_SC_CALL_TSX_DOES_NOT_EXIST, NULL, NULL, NULL); + ast_debug(3, "PJ_TRUE 1\n"); return PJ_TRUE; } else { if (ast_taskprocessor_alert_get()) { @@ -439,8 +440,8 @@ pjsip_rx_data_free_cloned(clone); } + ast_debug(3, "PJ_TRUE 3 - ready\n"); ast_taskprocessor_unref...

...n ignores the parameter and sends the fax at 9600. Grandstream doesn't, and all the faxes are going in and out at 2400. Looking at the code I found this in chan_sip.c (line 7736): if ((sscanf(a, "T38FaxMaxBuffer:%30u", &x) == 1)) { ast_debug(3, "MaxBufferSize:%d\n", x); found = TRUE; } else if ((sscanf(a, "T38MaxBitRate:%30u", &x) == 1) || (sscanf(a, "T38FaxMaxRate:%30u", &x) == 1)) { ast_debug(3, "T38MaxBitRate: %d\n", x); switch (x...

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

...erializer. >> */ >> pjsip_endpt_respond_stateless(ast_sip_get_pjsip_endpoint(), >> rdata, >> PJSIP_SC_CALL_TSX_DOES_NOT_EXIST, NULL, NULL, >> NULL); >> + ast_debug(3, "PJ_TRUE 1\n"); >> return PJ_TRUE; >> } else { >> if (ast_taskprocessor_alert_get()) { >> @@ -439,8 +440,8 @@ >> pjsip_rx_data_free_cloned(clone); >> } >> >> + ast_de...

...stream=<value optimized out>) at res_pjsip_t38.c:703 703 ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &sdp->conn->addr, sizeof(host)); (gdb) lsit Undefined command: "lsit". Try "help". (gdb) list 698 ast_debug(3, "Declining; T.38 state is rejected or declined\n"); 699 t38_change_state(session, session_media, state, T38_DISABLED); 700 return -1; 701 } 702 703 ast_copy_pj_str(host, stream->conn ? &stream->conn->addr : &a...

On Sun, Jun 11, 2017, at 01:31 PM, Michael Maier wrote: > On 06/11/2017 at 04:39 PM Joshua Colp wrote: > > On Sun, Jun 11, 2017, at 11:34 AM, Michael Maier wrote: > > > > <snip> > > > >>> > >>> PJSIP uses a dispatch model. The request is queued up, acted on, and > >>> then that's it. The act of acting on it removes it from

...t;= strlen(hostname) && strcasecmp(hostname+strlen(hostname)-strlen(str)+2, str+2) == 0) { ast_log(LOG_WARNING,"Warning: allowing match on wildcard (%s =~ %s)\n", hostname, str); ret = 0; } else { ret = -1; } ast_debug(3, "SSL %s compare s1='%s' s2='%s'\n", desc, hostname, str); OPENSSL_free(str); return ret; } -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230822/8ac31...

...>lock); pri->pvts[chanpos]->setup_ack = 1; /* Send any queued digits */ for (x = 0;x < strlen(pri->pvts[chanpos]->dialdest); x++) { ast_debug(1, "Sending pending digit '%c'\n", pri->pvts[chanpos]->dialdest[x]); pri_information(pri->pri, pri->pvts[chanpos]->call, pri->pvts[chanpos]->dialdest[x]); } ast_mutex_unlock(&pri->pvts[chanpos]->lock)...

Hi Looking at the source of app_voicemail.c there are many statements like: ast_debug(1, "%s doesn't exist, doing what we can\n", prefile); Where do I have to enably this to be showed in the console or logged to a file by logger. core set debug does not seem to help here. Well, my actual problem is, that if a customer has recorded his own greeting, he usualy tells...

I'm finally migrating from the 6/7/05 CVS version to 1.4. Had quite a run, I have to admit. Asterisk itself only segfaulted once or twice, but the dns issues have been bothering me. And the box just needs to go. Everything is going on a Ubuntu 6.06TLS server, that's been perfectly stable. I had 1.4.1 installed and running, but not configured. Yesterday I upgraded to 1.4.11,

Hi John This works fine providing extensions 1001,1002 and 1003 are "Incall" or "Paused" - the problem appears to be that is a handset say 1002 is "ringing" then the 2xxx then the penalty is not honoured. This is well described in the History section of the following link https://wiki.freepbx.org/display/PPS/lazymembers+patch+to+app_queue As I say this seems to

Hi, I have a problem with polarity reverse this my dahdi config [channels] context=default usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1

...>> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which can't be >> seen in tcpdump. >> >> I extended the above patch by adding the info at the last output: >> >> ast_debug(3, "PJ_TRUE 3 - ready %s\n", pjsip_rx_data_get_info(rdata)); >> >> This gives, that for *all* received packages return PJ_TRUE is reached. >> >> But: all packages besides of the phantom resend use the same address >> rdata0x7f963c0009b8 - only the phantom res...