search for: 00000007

...671f-8943-4773-9e36-398ef112a22f} a=rtcp-mux a=rtpmap:109 opus/48000/2 a=rtpmap:9 G722/8000/1 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=setup:actpass a=ssrc:1181629171 cname:{dc854b06-da58-45b3-8185-bbc6a57746c0} <-------------> --- (13 headers 31 lines) --- [May 19 16:47:43] DEBUG[14160][C-00000007]: chan_sip.c:25444 handle_request_invite: Initializing initreq for method INVITE - callid cf2990ba- 3f12-3d9e-adb6-52889c414ed3 Using INVITE request as basis request - cf2990ba-3f12-3d9e-adb6-52889c414ed3 Found peer 'vr1a882' for 'vr1a882' from 2.2.2.2:8558 [May 19...

...arted music on hold, class 'default', on SCCP/203-0000000c -- SCCP: Channel '11' freed by schedule. Output from Asterisk when it doesn't work: *CLI> -- SEP001121d89b97: (sccp_pbx_softswitch) New call on line 203 -- Executing [201 at home:1] Gosub("SCCP/203-00000007", "internal-defaults,1") in new stack -- Executing [internal-defaults at home:1] NoOp("SCCP/203-00000007", ""Lekrummet" <203>") in new stack -- Executing [internal-defaults at home:2] AGI("SCCP/203-00000007", "cid_lookup.agi&...

On 21-11-16 15:17, Matthew Jordan wrote: > > On Mon, Nov 21, 2016 at 7:05 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > when using Asterisk version 13.12.2 I notice that it takes up to > 30 seconds (sometimes even longer) for a call queue to call its > members. > >

...-> hylafax Same procedure: the receiving t38modem(2) sends ReInvite for t.38 - but this time, the extension / asterisk just ignores it. After the 5. retry to switch to T.38, asterisk tells: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007' => Why does asterisk reject the switch / ReInvite to T.38 this time? It never even tried to send it to the ISP! Thanks for any hint! Regards, Michael

...ace to enter the namespaces and security context of the container. But when I do the same thing in debian OS, It reported an error, with details as following: root@debian:/etc# vir list Id Name State ---------------------------------------------------- 4424 instance-00000007 running 25913 instance-00000008 running root@debian:/etc# vir dumpxml 4424 <domain type='lxc' id='4424'> <name>instance-00000007</name> <uuid>f1ce5360-bb5e-4cfc-b5ef-d05f8db52618</uuid> <memory unit='KiB'>...

...my dialplan '_5062XXXXXXX' => 1. Set(FAXOPT(t38gateway)=yes) [pbx_config] 2. Dial(DAHDI/g0/${EXTEN}) [pbx_config] 3. Hangup() [pbx_config] == Using SIP RTP CoS mark 5 -- Executing [50624309954 at termination-test:1] Set("SIP/robert-00000007", "GROUP(customers)=ibasis") in new stack -- Executing [50624309954 at termination-test:2] Set("SIP/robert-00000007", "GROUP(termination)=ss7") in new stack -- Executing [50624309954 at termination-test:3] Set("SIP/robert-00000007", "FAXOPT(...

On Fri, Jun 16, 2017, at 02:13 AM, Michael Maier wrote: > Has anybody any idea why asterisk drops the media stream in the 200 OK? > The channel has been T38_ENABLED before! Or is it necessary to add more > debug code? Who does the negotiating? > Only asterisk or is pjsip doing some parts, too? Asterisk does the T.38 negotiation and produces the answer SDP, PJSIP does the SDP

...;s what happens when external number 87133306 calls into my PRI, extension 1111 answers, does an attended transfer to 0102, and completes the transfer. The provider eventually hangs up with "Message not compatible with call state (101)". channel.c: Released clone lock on 'SIP/1111-00000007<ZOMBIE>' channel.c: Done Masquerading DAHDI/i1/87133306-3 (6) chan_dahdi.c: Requested indication 26 on channel DAHDI/i1/87133306-3 chan_dahdi.c: Requested indication 17 on channel DAHDI/i1/87133306-3 channel.c: Bridge stops because we're zombie or need a soft hangup: c0=SIP/1111-0000...

...ILED with error NT_STATUS_NO_SUCH_USER" in the winbindd log however I get a message that states "NTLM CRAP authentication for user [NZA]\[test-user] returned NT_STATUS_OK (PAM: 0)" Log snippit: [2005/09/19 13:41:16, 5] rpc_parse/parse_prs.c:prs_uint32(635) 0128 attr : 00000007 [2005/09/19 13:41:16, 5] rpc_parse/parse_prs.c:prs_uint32(635) 012c g_rid: 00002846 [2005/09/19 13:41:16, 5] rpc_parse/parse_prs.c:prs_uint32(635) 0130 attr : 00000007 [2005/09/19 13:41:16, 5] rpc_parse/parse_prs.c:prs_uint32(635) 0134 g_rid: 00002812 [2005...

Thanks Doug. Turns out if using hangup request does not work with the escaped character CLI> hangup request PJSIP/1003\ a-00000007 Usage: channel request hangup <channel>|<all> Request that a channel be hung up. The hangup takes effect the next time the driver reads or writes from the channel. If 'all' is specified instead of a channel name, all channels will see the hangup reque...

...EN}) same => n,Set(myNum=957) same => n,Set(sec=$[REMAINDER(${myNum},60)]) same => n,Set(sec=$[ABS(${sec})]) same => n,Set(sec=$[${MATH(${sec}+0,i)}]) same => n,Verbose(1,${myNum},${sec}) gives me -- Executing [7 at fromvoipfone201:1] Verbose("PJSIP/6001-00000007", "Context: fromvoipfone201 Exten:7") in new stack Context: fromvoipfone201 Exten:7 -- Executing [7 at fromvoipfone201:2] Set("PJSIP/6001-00000007", "myNum=957") in new stack -- Executing [7 at fromvoipfone201:3] Set("PJSIP/6001-00000007", "...

...7.3 -> "ge" works. #right If the rule modified to (spamtest :value "ge" :comparator "i;ascii-numeric" "6.0") everything works as expected. One more thing - sieve-test. I couldn't test my scripts with it: # sieve-test -x +spamtest -t sieve spam.txt 00000007: SPAMTEST test 00000007: spamtest: extension not configured 00000013: JMPFALSE (false) Performed actions: (none) Implicit keep: * store message in folder: INBOX Info: final result: success

...nel.c: DTMF end accepted with begin '#' on SIP/test13-0000000b [Jul 8 09:46:56] DTMF[22334] channel.c: DTMF end passthrough '#' on SIP/test13-0000000b [Jul 8 09:46:56] VERBOSE[22334] logger.c: [Jul 8 09:46:56] -- Started music on hold, class 'default', on SIP/test3-00000007 [Jul 8 09:46:56] VERBOSE[22334] logger.c: [Jul 8 09:46:56] -- <SIP/test13-0000000b> Playing 'pbx-transfer' (language 'be') [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begin '2' received on SIP/test13-0000000b [Jul 8 09:46:57] DTMF[22334] channel.c: DTMF begi...

On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > Just a guess (without knowing about your network), but are the two ends > points on public networks and visible to one another? If not the reinvite > may be passing an internal (nat'ed) address to the other and the connection > will fail...just a though t38modem -tt -o /var/log/t38modem.log --no-h323 -u 91 --sip-listen

On Mon, Jun 5, 2017, at 01:22 PM, Michael Maier wrote: > > Do you have any idea where to start to look at? Adding additional output > in the source code? Which functions could be interesting? I may add own > debug code to see why things are happening as they happen here. The logic for T.38 negotiation lives all in the res_pjsip_t38 module and the request to negotiate works using a

On 06/11/2017 at 06:51 PM Joshua Colp wrote: > On Sun, Jun 11, 2017, at 01:47 PM, Joshua Colp wrote: >> The distributor is in res/res_pjsip/pjsip_distributor.c, the distributor >> function being the entry point. That function returning PJ_TRUE >> indicates to PJSIP that it has been handled and no subsequent modules >> should be called by that running thread. The

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 the queue. > > That's the *expected* behavior ... . I rechecked again and again. All > existing tcpdumps. The "resent" package isn't part of

On Mon, Jun 5, 2017, at 12:00 PM, Joshua Colp wrote: > On Mon, Jun 5, 2017, at 11:49 AM, Michael Maier wrote: > > On 06/05/2017 at 11:30 AM, Joshua Colp wrote: > > > On Sun, Jun 4, 2017, at 10:40 AM, Michael Maier wrote: > > >> On 06/04/2017 at 01:41 PM Telium Technical Support wrote: > > >>> Just a guess (without knowing about your network), but are

Hi All, I have an issue on voice mail with odbc in asterisk 11.7 box. Voice message can be received through Google mail but it doesn't show in phone. The error messages is as follow and let me get your kind advice. -- <SIP/0015-00000007> Playing 'auth-thankyou.g722' (language 'en') [2014-05-28 14:55:13] DEBUG[12260][C-00000006]: app_voicemail.c:3824 last_message_index: Directory '/var/spool/asterisk/voicemail/default/701/INBOX' has no messages and therefore no index was retrieved. == Parsing '/var/...

On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote: <snip> > > t38modem and asterisk are using > > m=image 35622 udptl t38 > ^^^^^ > > Provider uses > > m=image 35622 UDPTL t38 > ^^^^^ > > Could this be a problem? If I'm sending internal only, it's always > lowercase. Looking at the tests we have we