search for: 6003

I have three video phones here for testing: Extension 6003 is Eyebeam Extension 6004 is a hard phone (model 8770) Extension 6005 is a hard phone (model 8882) Can anybody have a look at my settings and the output I get from all kinds of dialings, please. The sip settings for all phones is (user / password different): [6003] type=friend username=6003 sec...

Hello! I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") Thank you!

...upport' to 'yes' in sip.conf and set STURN and TURN server in rtp.conf. It still fails! Hope someone to help me out! Thanks in advance:) This is the output of CLI: ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ == Using SIP RTP CoS mark 5 -- Called SIP/6003 -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 is ringing -- SIP/6003-00000001 answered SIP/6004-00000000 -- Channel SIP/6004-00000000 joined 'simple_bridge' bas...

...ck == Spawn extension (from-sip, 9070168177, 3) exited non-zero on 'SIP/6002-9fac' The gatekeeper sees nothing from that. I guess the syntax is wrong for dialing. How should it be? Video connection: I try to call from an H323 soft phone through the gatekeeper to call the extension 6003 (eyebeam) H323 soft phone calls through GK Asterisk box without webcam installed: -- Executing Dial("H323/203.160.252.147-a44c", "SIP/8600") in new stack Jul 8 13:51:37 WARNING[12674]: chan_sip.c:1742 create_addr: No such host: 8600 Jul 8 13:51:37 NOTICE[12674]: app_dia...

...be doing a half duplex protocol conversion? Any insight would be greatly appreciated!! Current configuration: Fedora Core 1 Asterisk - 1.0.7 (had same problem on 1.0.6) SJPhone - 1.50.271d, Mar 11 2005 (WinXP) XLite - 1103m build stamp 14262 (WinXP) Zultys Zip2 - ZUTS 3.52 sip.conf exerpt: [6003] ; (A) type=friend regexten=6003 username=6003 host=dynamic disallow=all ;allow=gsm allow=ulaw [6004] ; (C) type=friend regexten=6004 username=6004 host=dynamic disallow=all ;allow=gsm allow=ulaw [2101] ; (B) type=friend regexten=2101 username=2101 host=dynamic disallow=all ;allow=gsm allow...

...files, which has caller id in them, to call >> originate from dial plan. >> But I don't see such parameter here >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate >> >> How can I pass callerid to following: >> >> exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") > > I use this patch > > https://messinet.com/rpms/browser/asterisk/asterisk-12-app_originate_callerid.patch Thank you! I'll try it. > > because of https://issues.asterisk.org/jira/browse/ASTERISK-23016 Unfortun...

I'm in the process of testing a TLS/SRTP install. My experience is improving with each new challenge, but this one is a great test of my 2 month experience with Asterisk. When I dial 6003 from 6001, it takes 35 seconds until I get the error message that 6003 is circuit-busy. Any help would greatly be appreciated. Below is the error message and the extensions and sip.conf files. *CLI> == Using SIP RTP CoS mark 5 -- Executing [6003 at myphones:1] Set("SIP/6001-00000...

...class Report < ActiveRecord::Base has_and_belongs_to_many :region has_and_belongs_to_many :subjects has_and_belongs_to_many :sorts end And i would like to get a count like @sort.reports.count The problem is get this query: SELECT COUNT(*) FROM reports WHERE (reports_sorts.sort_id = 6003 ) He uses the wrong table "reports" instead of the join table "reports_sorts". But it is in the where clause. Thanks in advance, Martijn

...am = "/home/xxx/scripts/mongrel_rails_start 6002" stop program = "/home/xxx/scripts/mongrel_rails_stop 6002" if totalmem > 50.0 MB for 5 cycles then restart if failed port 6002 protocol http with timeout 10 seconds then restart group mongrel #6003 check process mongrel-6003 with pidfile /home/xxx/sshare/app/log/mongrel.6003.pid start program = "/home/xxx/scripts/mongrel_rails_start 6003" stop program = "/home/xxx/scripts/mongrel_rails_stop 6003" if totalmem > 50.0 MB for 5 cycles then restart if failed...

...yMode Access R/W Oplock SharePath Name Time ---------------------------------------------------------------------------- ---------------------- 10313 6011 DENY_NONE 0x81 RDONLY NONE /home/web data/clients/canadians Tue Nov 21 12:35:35 2006 10295 6003 DENY_NONE 0x81 RDONLY NONE /home/web data/clients/quailelectronics Thu Nov 23 13:03:44 2006 10295 6003 DENY_NONE 0x81 RDONLY NONE /home/web data/clients/quailelectronics Fri Nov 24 08:10:29 2006 10295 6003 DENY_NONE 0x81 RDONLY...

We come into this section of the dialplan: exten => 88670333333,1,Wait(1) exten => 88670333333,n,SayUnixTime exten => 88670333333,n,NoOp(If you know the extension ...) exten => 88670333333,n,Dial(${PHONE_6003}) The caller from the GK hears only ringing, not the time. The extension 6003 rings and I can pick up, but without any voice nor video. athome*CLI> -- Executing Wait("H323/GVSC8770-e822", "1") in new stack -- Executing SayUnixTime("H323/GVSC8770-e822", &q...

...is the trace: Frame 96 (808 bytes on wire, 808 bytes captured) Session Initiation Protocol Request-Line: INFO sip:6002@192.168.10.90 SIP/2.0 Method: INFO Resent Packet: False Message Header Call-ID: 60b8596c-4135c-c0a81e68@192.168.10.90 From: Demo2<sip:6003@192.168.10.90;user=phone>;tag=221a0-a1cf SIP Display info: Demo2 SIP from address: sip:6003@192.168.10.90 SIP tag: 221a0-a1cf To: <sip:6002@192.168.10.90;user=phone>;tag=as6b294484 SIP to address: sip:6002@192.168.10.90 SI...

...GIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Are hints not yet implemented in res_jabber? I have this here: exten => 3000,hint,gtalk/gtalk_account/mari.xxxxxxx at gmail.com But the hint doesn't show any difference. It always shows online on the phone and core show hints always shows that: 6003 at internal : SCCP/6003 State:Unavailable Watchers 0 6002 at internal : SCCP/6002 State:Idle Watchers 0 6001 at internal : SCCP/6001 State:Idle Watchers 0 6000 at internal : SCCP/6000 State:Idle Watchers 0 6004 at internal : SIP...

...es on wire, 808 bytes captured) > Session Initiation Protocol > Request-Line: INFO sip:6002@192.168.10.90 SIP/2.0 > Method: INFO > Resent Packet: False > Message Header > Call-ID: 60b8596c-4135c-c0a81e68@192.168.10.90 > From: Demo2<sip:6003@192.168.10.90;user=phone>;tag=221a0-a1cf > SIP Display info: Demo2 > SIP from address: sip:6003@192.168.10.90 > SIP tag: 221a0-a1cf > To: <sip:6002@192.168.10.90;user=phone>;tag=as6b294484 > SIP to address: sip:6002@192...

Full_Name: Sundar Dorai-Raj Version: 1.8.1 OS: Windows 2000 Professional Submission from: (NULL) (12.64.199.173) I discovered this problem when trying to use princomp in package:mva when a column in my matrix was all zeros and I set cor = TRUE (thus division by 0). Doing so hangs R, never to return. I have to shut down Rterm in the Task Manager and lose all work from the current image. I tracked

...ullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = SECRET threewaycalling = yes registeriax = no registersip = yes canreinvite = no nat = no dtmfmode = rfc2833 Queues.conf [6003] fullname = All of us strategy = ringall timeout = wrapuptime = autofill = yes autopause = no maxlen = joinempty = no leavewhenempty = no reportholdtime = no musicclass = member = Agent/6001 member = Agent/6002 extensions.conf - broken [DID_trunk_2] include = default exten = _X....

...n looks something like this: sip.conf: [mjg] type=friend username=mjg context=sip callerid="Masuo" <6001> secret=**** host=dynamic defaultip=199.242.227.227 canreinvite=no mailbox=6001 subscribecontext=sip [pjf] type=friend username=pjf context=sip callerid="Patrick" <6003> secret=**** host=dynamic defaultip=199.242.227.227 canreinvite=no mailbox=6003 subscribecontext=sip 360 configuration: fkey6!: dest &lt;sip:6001@voip.eti.wips.gov;user=phone&gt; fkey7!: dest &lt;sip:6003@voip.eti.wips.gov;user=phone&gt; extensions.conf: [macro-oneline] exten =...

...ten,7623,sip/7623&sip/7624") in new stack -- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008", "sip/7623&sip/7624&iax2/7623,20,t") in new stack == Using SIP RTP CoS mark 5 -- Called 7623 == Using SIP RTP CoS mark 5 [Apr 4 12:46:37] WARNING[6003]: acl.c:698 ast_ouraddrfor: Cannot connect [Apr 4 12:46:37] WARNING[6003]: chan_sip.c:3115 __sip_xmit: sip_xmit of 0x8379240 (len 787) to 0.0.29.200:5060 returned -1: Invalid argument -- Called 7624 -- Called 7623 -- SIP/7623-00000009 is ringing [Apr 4 12:46:38] WARNING[5982]: chan_si...

...e (SIP) : 200 993266: Dec 16 14:53:23: ISDN Se7/0:1:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x80F3 And here is the output when the Asterisk box tries to make outbound calls through the 5400: 993314: Dec 16 14:57:38: //-1/909192939899/DPM/ dpAssociateIncomingPeerCore: Calling Number=6003, Called Number=96519590, Voice-Interface=0x0, Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE, Peer Info Type=DIALPEER_INFO_SPEECH 993315: Dec 16 14:57:38: //-1/909192939899/DPM/ dpAssociateIncomingPeerCore: Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Di...

..... the echo seems to be about 0.3 seconds delayed to the speech .. there is no echo on incoming voice, just an echo of my own voice as I speak. 2nd question: using a grandstream phone & asterisk, if I hear another phone ringing, how can answer it from the phone infront of me? eg. if extension 6003 is ringing, and i have phone number 6004, how can I answer it ? 3rd question: can someone give me some "starter hints" to configure call parking ? I haven't managed to find a direct way to transfer a call from phone to phone except using blind transfer and I want the person initiatin...