search for: xxxxxxxxxxx

Hello ! In order to chase after a problem I implemented the following dialplan to have an answertime of exactly one minute: exten => xxxxxxxxxxx,1,NoOp(Test wait) exten => xxxxxxxxxxx,n,Answer exten => xxxxxxxxxxx,n,NoOp(Current timestamp: ${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}) exten => xxxxxxxxxxx,n,Wait(60) exten => xxxxxxxxxxx,n,NoOp(Current timestamp: ${STRFTIME(${EPOCH},,%C%y%m%d%H%M%S)}) exten => xxxxxxxxxxx,n,Hangup B...

Return-path: <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx> Envelope-to: xxxxx at xxxxxxxxx Delivery-date: xxx, xx xxx xxxx xx:xx:xx +xxxx Received: xxxx [xxx.x.x.x] (xxxx=xxxxxxxxx) xx xxxxxxxxx.xxxxxxxxxxxx.xx xxxx xxxxx (xxxx x.xx) (xxxxxxxx-xxxx <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx>) xx xxxxxx-xxxxxx-xx xxx xxxxx at xxxxxxxxx; xxx, xx xxx xxxx xx:xx:xx +xxxx Delivered-To: xxxxxxx.xxxxxxx at xxxxx.xxx Received: xxxx xxxxx-xxxx.x.xxxxxx.xxx [xx.xxx.xxx.xx...

...rsion="1.0" encoding="UTF-8"?> <Container> <id>aQlfVHX+qPM</id> <lifetime>2009-09-19T08:14:55Z</lifetime> </Container> The response should contain the next headers: Content-Type=`application/vnd.3gpp+xml` Authentication-Info={rspauth=`xxxxxxxxxxx...xxxxxxxxxxx`,qop=`auth-int`,cnonce=`xxxxxxxxxxxxxxxxxx....xxxxxxxxxxxxx`,nc=`000 00001`} How can I do that? I always see {"Cache-Control"=>"no-cache", "Content-Type"=>"application/xml"} when I print the headers... I can use xml builder ok to genera...

...on the phone shows 'Session Progress (in 183)' for the duration of the call, rather than 'Connected', so it looks like the SIP phone is not recognising call connection on the PSTN. Output from the console is as follows: -- Executing Dial("SIP/ronan-5e0e", "Zap/4/xxxxxxxxxxx") in new stack -- Called 4/xxxxxxxxxxx -- Hungup 'Zap/4-1' == Spawn extension (default, xxxxxxxxxxx, 1) exited non-zero on 'SIP/ronan-5e0e' A packet trace from the * box shows: ... 16.758516 192.168.2.9 -> 192.168.2.30 UDP Source port: 12230 Destination port...

...xxxxxxxx at xxxxxx.xxxxxx.xxxxx.xx.xx Tue Jun 03 09:14:33 2008 Message-ID: <xxxxxxxx.xxxxxxx at xxxxxx.xxxxx.xx.xx> X-UID: 3913 Status: RO X-Keywords: Content-Length: 6817 xxxx: xxx, xx xxx xxxx xx:xx:xx +xxxx xxxx: xxxxxxx xxxxxxxx <xxxxxxx.xxxxxxxx at xxxxxx.xxxxx.xx.xx> xxxx-xxxxx: xxxxxxxxxxx x.x.x.x (xxxxxxx/xxxxxxxx) xxxx-xxxxxxx: x.x xx: "xxxxxxxx at xxxxxxxx.xx" <xxxxxxxx at xxxxxxxx.xx> xx: xxxxx.xxxxxxxxxx at xxxxxxxxxxxx.xx, xxxxxx xxxxx <xxxxxx.xxxxx at xxxxxxxx.xxxxxx.xx>, xxxxxxx xxxxxxxxxxx <xxxxxxxxxx at xxxxxxxxxxx.xxx> xxxxxxx: xx: x: xx: xxx...

...WriteFormat: 0x2 (gsm) ReadFormat: 0x2 (gsm) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 111 Frames in: 6199 Frames out: 4847 Time to Hangup: 0 Elapsed Time: 3h29m16s Direct Bridge: <none> Indirect Bridge: <none> -- PBX -- Context: XXXXXXXXXXX Extension: XXXXX Priority: XXXXXX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNT...

...sterisk> on the email notification, and the message envelope simply doesn't say anything (does not actually play the vm-unknown message). So, following the tip provided by several previous posters, I tried putting this in my extensions.conf (the xx's are my DID, obviously: exten => xxxxxxxxxxx,1,SetCallerID(<Unknown ID>); exten => xxxxxxxxxxx,2,Goto(veeport_context,s,1); Okay. This works find for BLOCKED callerids. But the problem is that now all callerids are being treated as blocked, not just the really blocked ones!Help. What I want to do is: For calls that don't ha...

...t times, and I believe it may be causing issues of cross lines (an outgoing call gets mixed with an incoming call, both ending up on the same line). Incoming calls are processed fairly quickly, about 3 seconds which is perfectly acceptable. [Oct 2 10:30:27] DEBUG[22199] chan_zap.c: Dialing 'xxxxxxxxxxx' [Oct 2 10:30:27] DEBUG[22199] chan_zap.c: Deferring dialing... [Oct 2 10:30:27] VERBOSE[22199] logger.c: -- Called 4/xxxxxxxxxxx [Oct 2 10:30:35] DEBUG[22199] chan_zap.c: Engaged echo training on channel 4 [Oct 2 10:30:38] DEBUG[22199] chan_zap.c: Echo cancellation already on [Oct 2 1...

Recently, I got the following error messages in CLI periodically. Jan 20 17:43:18 ERROR[8641]: chan_sip.c:11002 handle_request_subscribe: Got SUBSCRIBE for extension XXXXXXXXXXX@from-int from 192.168.0.123, but there is no hint for that extension I have no idea what the error message tell me. I am sure I haven't that account XXXXXXXXXXX in my database and there is no hint extensions in dial plan as well. Anyone can help tell me why there is such an error message? a...

...k_3}/${ARG1}) exten => s,204,Hangup() [from-internal] exten => _NXXNXXXXXX,1,Macro(trunkdial,+1${EXTEN}) exten => _1NXXNXXXXXX,1,Macro(trunkdial,+${EXTEN}) sip.conf: [trunk1] host=xxx.xxx.xxx.xxx port=5060 type=peer allow=ulaw dtmfmode=rfc2833 canreinvite=no reinvite=no nat=no fromuser=+xxxxxxxxxxx call-limit=1 [trunk2] host=xxx.xxx.xxx.xxx port=5060 type=peer allow=ulaw dtmfmode=rfc2833 canreinvite=no reinvite=no nat=no fromuser=+xxxxxxxxxxx call-limit=1 [trunk3] host=xxx.xxx.xxx.xxx port=5060 type=peer allow=ulaw dtmfmode=rfc2833 canreinvite=no reinvite=no nat=no fromuser=+xxxxxxxxxxx cal...

Has anyone configured ast_fax (sending faxes via asterisk) with sendmail? The creation of rules to trap all numbers <mailto:XXXXXXXXXXX@domain.com> XXXXXXXXXXX@domain.com seems too complicated. Does anyone have setup details to share? (I don't want to switch MTA's). As a workaround, I could launch the app automatically from sendmail using an alias like: fax: "| /ast_fax" That way sending to <mai...

...only one server. A command as simple as "smbclient -U user //SERVER/SHARE", if fired up several times in a row on the server itself, will sometimes succeed, somtimes fail. Same happens for "smbclient -U root -L //SERVER". A sample log: > smbclient -d 5 -U xxxxxx //XXXXX/xxxxxxxxxxx > INFO: Current debug levels: > all: 5 > tdb: 5 > printdrivers: 5 > lanman: 5 > smb: 5 > rpc_parse: 5 > rpc_srv: 5 > rpc_cli: 5 > passdb: 5 > sam: 5 > auth: 5 > winbind: 5 > vfs: 5 > idmap: 5 > quota: 5 > acls: 5 &gt...

...V2 V3 V4 1 86.2700 49.9380 30.7630 0.1327 2 89.5127 55.9707 33.7683 0.1186 3 91.1833 58.4670 34.5610 0.1134 matplot(t, pch = 1:4, type = "o", col = rainbow(ncol(t)),xlab = "No. of inputs ", ylab = "Mean of 6 Datasets", main = "xxxxxxxxxxx") Instead of have typical x axes' range as "1, 2, 3, and 4" for each column, I need to indicate "11 Vars, 10 Vars, 6Vars, and 4Vars" or "A, B, C, and D" Does anybody know anything about it? I appreciate if you could help me out. Thank you very much indeed....

...2BZs-3D *See attached cloud infrastructure diagram for more details.* As a Cloud provider we are using Google Cloud Platform. Servers Specs GlusterFS nodes 3 Nodes each one with 2 Vcore 4 GB ram 100 SSD external disk (Gluster bricks) We were able to setup and start a GlusterFS volume: Volume Name: XXXXXXXXXXX Type: Replicate Volume ID:XXXXXXXXXXXXXXXXXXXXX Status: Started Snapshot Count: 0 Number of Bricks: 1 x 3 = 3 Transport-type: tcp Bricks: Brick1: node01:/mnt/disks/brick01/XXXXXXXXXXX Brick2: node02:/mnt/disks/brick02/XXXXXXXXXXX Brick3: node03:/mnt/disks/brick03/XXXXXXXXXXX Options Reconfigured: t...

...-- Hungup 'Zap/1-1' I also attempted a call from one of my POTS lines back into the other and got: *CLI> -- Executing ChanIsAvail("SIP/100-e429", "Zap/2") in new stack -- Hungup 'Zap/2-1' -- Executing Dial("SIP/100-e429", "Zap/2-1/XXXXXXXXXXX") in new stack WARNING[213006]: File chan_zap.c, Line 5591 (zt_request): Unknown option '-' in '2-1/XXXXXXXXXXX' -- Called 2-1/XXXXXXXXXXX -- Zap/2-1 answered SIP/100-e429 -- Starting simple switch on 'Zap/1-1' -- Executing Answer("Zap/1-1", &q...

...ry D-Channel on span 1 up for TEI 64 -- Accepting overlap voice call from '' to '<unspecified>' on channel 0/2, span 1 -- Starting simple switch on 'Zap/2-1' -- Executing Dial("Zap/2-1", "IAX2/<user>:<password>@sip.coco-connect.de/XXXXXXXXXXX") in new stack -- Called <user>:<password>@sip.coco-connect.de/XXXXXXXXXXX -- Call accepted by 62.180.50.221 (format g729) -- Format for call is g729 -- Channel 0/2, span 1 got hangup -- Hungup 'IAX2/62.180.50.221:4569/1' == Spawn extension (extern, XX...

.... after taking debug trace of both methods what I found was that a SIP HEADER parameter "rinstance" was missing in "to" and "INVITE" header fields when dialing via IP:PORT. I think this parameter is assigned automatically by asterisk. *NORMAL DIAL * INVITE sip:XYZ at xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588 SIP/2.0 To: <sip:XYZ at xxxxxxxxxxxx:28614;rinstance=0266b8b94f488588> Contact: <sip:1334225544 at xxxxxxxxxxx:5060> *IP DIAL* INVITE sip:XYZ at xxxxxxxxxxx:28614 SIP/2.0 To: <sip:XYZ at xxxxxxxxxxxx:28614> Contact: <sip:1334225544 at xxxxxxxx...

Hello Folks; I have an Asterisk server Asterisk 11.7.0 built by root @xxxxxxxxxxx on a x86_64 running Linux on 2013-12-27 18:47:44 UTC No FreePBX, no AsteriskNOW, no Elastix. Just Asterisk. Is there an API out there that anyone knows of that I can pass commands, etc to Asterisk? Creating Extensions, adding voicemail users, setting up voicemail, etc? I'm kind of clueless....

Hi, I have a IAX2 trunk between two sites (connected with an high bandwidth link) but sometime/often I get: chan_iax2.c: Auto-congesting call due to slow response and call is dropped (and routed on a PSTN link). In iax.conf, I have: [iax-out] username=iax-in type=peer trunk=yes secret=xxxxxxxxxxx qualify=yes host=xxx.yyy.zzz.32 auth=md5 Any idea? Perpaphs is due to 'qualify=yes'... Thanks -- Domenico Viggiani

Has anybody out there tried to use two different iconnecthere accounts with Asterisk? What I want to do is use a second account if the first is busy. I have tried the following: exten=>_91NXXNXXXXXX,1,StripMSD,1 exten=>_1NXXNXXXXXX,2,Dial,SIP/BYEXTENSION@iconnect ;iconnect is the first account exten=>_1NXXNXXXXXX,3,Dial,SIP/BYEXTENSION@iconnect2 ;iconnect2 is the second account But that