search for: res_rtp_asterisk

Hello, What does it mean when "rtp set debug ip" shows RTP packets that have been send, but there is no audio ? There was no audio on my call in both directions, but "rtp set debug" shows that there were RTP packets send. There is no firewall active on my Asterisk server : [root at sip asterisk]# /sbin/service iptables status iptables: Firewall not running. Kind

...ng g722 and g711 and gets exactly this invite back as incoming call. The answer is g722,g711 in the ok sdp. Now, Asterisk can't decide, which codec to use. It frequently changes the codec just as it likes to apparently without any visible reason. [2017-05-11 17:28:03] DEBUG[5121][C-0000003a]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-11 17:28:03] DEBUG[5113][C-00000039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from none to alaw [2017-05-11 17:28:04] DEBUG[5123][C-00000039]: res_rtp_asterisk.c:3634 ast_rtp_write: Ooh, format changed from non...

Hello, 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, I can't compile asterisk having error    [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka ‘struct pj_ice_sess_cb’} has no member named ‘on_valid_pair’   .on_valid_pair = ast_rtp_on_valid_pair,    ^~~~~~~~~~~~~ res_rtp_asterisk.c:2674:19: warning: initialization of ‘void (*)(pj_ice_sess *, pj_status_t)...

...are/info --with-misdn --with-sounds-cache=no --with-srtp --with-ssl=/opt/openssl101/usr --with-crypto=/opt/openssl101/usr Note the --with-ssl and --with-crypto options at the end, pointing to my openssl directory. After this I compile, but I am getting these messages when compilation reaches res/res_rtp_asterisk.c: a - output/pjlib-x86_64-redhat-linux-gnu/sock_qos_bsd.o a - output/pjlib-x86_64-redhat-linux-gnu/ssl_sock_common.o a - output/pjlib-x86_64-redhat-linux-gnu/ssl_sock_ossl.o a - output/pjlib-x86_64-redhat-linux-gnu/ssl_sock_dump.o a - output/pjlib-x86_64-redhat-linux-gnu/string.o a - output/pjlib...

...been running a minimal (ish) SIP based Asterisk with the modules based on chan-sip. For various reasons unrelated to Asterisk the machine the latest incarnation of this configuration has been updated to Debian Buster and thus to Asterisk 16. Since this upgrade I have a dependency problem related to res_rtp_asterisk.so. So the old config was: [modules] autoload=no load => res_rtp_asterisk.so load => res_http_websocket.so load => chan_local.so load => codec_ulaw.so load => codec_alaw.so load => pbx_config.so load => chan_sip.so load => app_dial.so load => func_callerid.so load =>...

Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] WARNING[15202]...

...3, 2020 at 9:27 AM Administrator <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > Hello, > > 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of > them, > I can't compile asterisk having error > >     [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o > res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka ‘struct > pj_ice_sess_cb’} has no member named ‘on_valid_pair’ >    .on_valid_pair = ast_rtp_on_valid_pair, >     ^~~~~~~~~~~~~ > res_rtp_asterisk.c:2674:19: warning: initializati...

...ng functionality into Asterisk 12.0.0 ? features.conf: parkswitch => *#,callee/caller,Macro(parkswitch) extensions.conf: [default] .... include => parkedcalls [macro-parkswitch] exten => s,1,ParkAndAnnounce(,,PARKED,SIP/100) messages: [Jan 30 21:00:00] DEBUG[7114][C-00000000]: res_rtp_asterisk.c:2847 create_dtmf_frame: Creating BEGIN DTMF Frame: 42 (*), at x.x.x.x:9530 [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:4050 __ast_read: DTMF begin '*' received on SIP/at-tcty-ssw-00000000 [Jan 30 21:00:00] DTMF[7114][C-00000000]: channel.c:4061 __ast_read: DTMF begin passthroug...

Hi, compiling the current git version on Centos 7 gives me: [CC] res_statsd.c -> res_statsd.o res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified in initializer .on_valid_pair = ast_rtp_on_valid_pair, ^ res_rtp_asterisk.c:2669:2: warning: initialization from incompatible pointer type [enabled by default] res_rtp_asterisk.c:2669:2: warning: (near initialization for ‘ast_rtp_ice_sess...

On Fri, Mar 13, 2020 at 9:27 AM Administrator <admin at tootai.net> wrote: > Hello, > > 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, > I can't compile asterisk having error > > [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o > res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka ‘struct > pj_ice_sess_cb’} has no member named ‘on_valid_pair’ > .on_valid_pair = ast_rtp_on_valid_pair, > ^~~~~~~~~~~~~ > res_rtp_asterisk.c:2674:19: warning: initialization of ‘void > (*)...

I have been using 11.4.0 for some time. All was fine. I downloaded 11.5, extracted, run ./configure, make, make install. I got a message about res_rtp_asterisk.so was not compiled in the 11.5 Sure enough I have rss_rtp_asterisk.c but not .o file and no .so file. I then looked in the config.log and nothing is in there about res_rtp_asterisk What's up? jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://l...

...CT_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 to: xx.xx.154.111:18578 [Feb 22 11:17:00] VERBOSE[29023][C-00024cd5] res_rtp_asterisk.c: > 0x2b315c01cbc0 -- Strict RTP learning after remote address set to: xx.xx.0.12:16498 [Feb 22 11:17:00] VERBOSE[28191][C-0...

...s(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 to: > xx.xx.154.111:18578 > [Feb 22 11:17:00] VERBOSE[29023][C-00024cd5] res_rtp_asterisk.c: > > 0x2b315c01cbc0 -- Strict RTP learning after remote address set to: > xx.xx.0.12:16498 > [Feb 2...

...(alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|g729) [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] res_rtp_asterisk.c: 0x7f02241bdaf0 -- Strict RTP learning after remote address set to: 41.11.11.11:13918 [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Peer audio RTP is at port 41.11.11.11:13918 [2020-09-19 23:42:19] VERBOSE[2637][C-00021a1f] chan_sip.c: Looking for 0100000000 in from-pstn (domain 52....

...n SDP renegotiation in the middle of a call, and seemingly as a consequence Asterisk re-locked on the source and destination servers...but also registered third server 10.0.0.125. This seems to have broken the call to the desired destination server. [2020-01-14 18:08:25] VERBOSE[29350][C-00000006] res_rtp_asterisk.c: 0x7f40240322e0 -- Strict RTP switching source address to 10.0.0.228:42150 [2020-01-14 18:08:26] VERBOSE[29324][C-00000006] res_rtp_asterisk.c: 0x7f403c00c3b0 -- Strict RTP learning complete - Locking on source address 10.0.0.192:22522 [2020-01-14 18:08:26] VERBOSE[29350][C-00000006] res_rtp_aste...

I have a CentOS 6.3 machine I installed Asterisk 11, worked fine... I then tried to install on Cents 5.8, seemed to go fine... Then when I placed a call I got this: ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? Did a search and found issues with ARM and this problem but did not help me, not using gtalk or anything. Just call between two polycom phones on local network.

Hello ! My asterisk log is full of messages like this: [2011-03-06 19:01:15] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:20] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:25] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:30] DEBUG...

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 or a maximum? The unusual call flow in question results in > Asterisk learni...

...in 1762250 allocations in file taskprocessor.c 26125344 bytes in 203171 allocations in file rtp_engine.c 17308827 bytes in 307848 allocations in file stasis_cache.c 9548128 bytes in 35482 allocations in file stasis_bridges.c 3923172 bytes in 92169 allocations in file res_rtp_asterisk.c 3099771 bytes in 29185 allocations in file strings.c Next minute it was already like this: # cat memory_show_summary_201711101145.log | sort -nr | head 3091009032 bytes allocated (375561 in caches) in 16345842 allocations 1482476997 bytes in 1663899 allocations in file stasis_channels...

I'm attempting to find where my extra long DTMF Tones are coming from. I'm dialing from my sip handset through my proxy to my Asterisk box which is my PSTN Gateway. I'm pressing 4 to select a menu and everything is fine. [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin '4' received on SIP/trunk-0a02dee0 [Feb 12 16:58:18] DTMF[29762] channel.c: DTMF begin passthrough