search for: 4577

...logged off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing AMD("Local/91441425477394 at default-b9f2,1", "2000|2000|1000|5000|120|50|4|256") in new stack -- AMD: Local/91441425477394 at default-b9f2,1 0000000000 (null...

...in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 -- Executing AGI("Local/192*168*000*002*78600051 at default-230b,2", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("Local/192*168*000*002*78600051 at default-230b,2", "SIP/192*168*000*002*78600051 at sip64||tTor") in new stack -- Called 192*168*000*002*78600051 at sip64 == Spawn...

...er center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works perfectly. Normal Hang Up : ----------------------------- Quote: vici*CLI> -- Executing AGI("SIP/cc101-09f44300", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-09f44300", "SIP/VOIP74/17274507674||tTor") in new stack -- Called VOIP74/17274507674 == Manager 'sendcron' logged off from 127.0.0.1 -- SIP/VOIP74...

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack -- Called Sama203/119545090201 Sep 8 14:19:09 WARNING[2813]: chan_sip.c:9890 handle...

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38 WARNING[2884]: chan_oss.c:585 setformat: Unable to re-open...

...ly destroying 1, having received hangup Sep 27 16:20:36 VERBOSE[10049] logger.c: == Spawn extension (default, 8600051, 1) exited non-zero on 'IAX2/172.22.0.5:4569-1' Sep 27 16:20:36 VERBOSE[10049] logger.c: -- Executing DeadAGI("IAX2/172.22.0.5:4569-1", "agi://127.0.0.1:4577/call_log") in new stack Sep 27 16:20:36 VERBOSE[10049] logger.c: -- AGI Script agi:// 127.0.0.1:4577/call_log completed, returning 0 Sep 27 16:20:36 VERBOSE[10049] logger.c: -- Executing DeadAGI("IAX2/172.22.0.5:4569-1", "agi:// 127.0.0.1:4577/VD_hangup--HVcauses--PRI---...

...n conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.27 currently running on vicidialnow (pid = 2615) Verbosity is at least 21 -- Executing AGI("SIP/cc101-b7a07420", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7a07420", "SIP/17275691533 at sip8||tTor") in new stack -- Called 17275691533 at sip8 -- SIP/sip8-0825f9b0 is making progress passing it to SIP/cc101-b7a...

I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ;

I am getting this Error on my Asterisk. How to solve it ? "ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090115/cb953962/attachment.htm

Hi all, I have n no. of extensions in my dialer. from 456 to 556 extensions. I was created 2 other extensions 667 and 668 I need to allow only STD calls to go from this extensions. These all extensions are same context . I need to define the STD dialplan for only this 2 extensions. how I can ? Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI |

...5060 Unmonitored 3210/3210 192.168.0.112 D 5060 Unmonitored 4 sip peers [4 online , 0 offline] -- Executing Ringing("SIP/3210-084eaa80", "") in new stack -- Executing AGI("SIP/3210-084eaa80", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/3210-084eaa80", "SIP/3213)|30|to") in new stack Feb 3 12:42:25 WARNING[10368]: chan_sip.c:1994 create_addr: No such host: 3213) Feb 3 12:42:25 NOTICE[...

...BRI card showing OK with dahdi_tool, this NT mode. whenever I am dialing from server i am not able to connect the call . in Cli below mention warning is comming . please what is the mistake with me . help me Executing [0559566768 at default:1] AGI("Console/dsp", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing [0559566768 at default:2] Dial("Console/dsp", "Dahdi/g0/0559566768|55|tTo") in new stack [May 31 01:10:09] WARNING[30356]: channel.c:3443 ast_request: No channel...

...rd 0 Span 4" (MASTER) AMI/CCS RED span=4,4,0,ccs,ami # termtype: te bchan=10-11 dchan=12 echocanceller=mg2,10-11 # Global data loadzone = us defaultzone = us [root at go asterisk]# extensions.conf TRUNK=Dahdi/g0 [default] exten => _0XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _0XXXXXXXXX,2,MixMonitor(/var/spool/asterisk/astrec/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}-${CALLERID(num)}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten => _0XXXXXXXXX,3,Dial(${TRUNK}/${EXTEN},,tTo) exten => _0XXXXXXXXX,4,Hangup exten => _XXXXXXXXX,1,AGI(agi://127.0.0.1:4...

...; Trunk interface TRUNKX=Zap/g2 ; 2nd trunk interface ... ... ; dial a long distance outbound number to SPAIN ; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls, exten => _00034XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _00034XXXXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},55,To) exten => _00034XXXXXXXXX,3,Hangup ... ... and it works fine, but I need to start working with my second span & I don't know how to add it in extensions.conf file. -------------- next part -------------- An HTML attac...

IVR Number :17275691533 When I try it from xlite configuring my provider directly, it works perfectly. When I try to dial out from dialer , it doesnt work. [sip8] type=peer username=user fromuser=user authuser=user secret=password host=8.14.146.111 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 What cld be the reason ? --------------

...00 DID's which is 4578900 to 4578999 , and i have 5001 to 5099 extensions. when incomming call come to this DID no. (4578901) that time 5001 extestinsion should ring. below my dial plan is not getting any result , inthat has any mistakes. please help. exten => _45789XX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _45789XX,1,Set(Dest=2{EXTEN:-2}) exten => _45789XX,2,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten => _45789XX,3,Dial(SIP/${Dest},,tTo) exten => _45789XX,4,Hangup Best Regards, Mahesh Katta *BUZZ**WORKS* B...

...ten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)}) exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2}) exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident}) exten => _0XXXXXXXXX,4,Set(CALLERID(name)=${outgoing_ident}) exten => _0XXXXXXXXX,5,AGI(agi://127.0.0.1:4577/call_log) exten => _0XXXXXXXXX,6,Set(CALLERID(num)=${outgoing_ident}) exten => _0XXXXXXXXX,7,MixMonitor(/var/spool/asterisk/astrec/${TIMESTAMP}-${CALLERIDNUM}-${EXTEN}-${UNIQUEID}.gsm|av(0)V(0)) exten => _0XXXXXXXXX,8,Dial(${TRUNK}/${EXTEN},,tTo) exten => _0XXXXXXXXX,9,Hangup this dia...

...ull: Had to drop call because I couldn't make SIP/cc101-b790c1d8 compatible with SIP/sip19-090e87d8 == Spawn extension (default, 719545090000, 2) exited non-zero on 'SIP/cc101-b7 90c1d8' -- Executing DeadAGI("SIP/cc101-b790c1d8", "agi:// 127.0.0.1:4577/call_log--H Vcauses--PRI-----NODEBUG-----16---------------") in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1 6--------------- completed, returning 0 -------------- n...

...slave 65 from 62 [Oct 29 11:44:47] DEBUG[12424]: chan_dahdi.c:1630 conf_del: Removed 77 from conference 3848/1023 [Oct 29 11:44:47] DEBUG[12424]: chan_dahdi.c:1630 conf_del: Removed 74 from conference 3848/1023 -- Executing [h at from-pstn:1] DeadAGI("Zap/62-1", "agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----2") in new stack [Oct 29 11:44:47] WARNING[12424]: res_agi.c:2203 deadagi_exec: Running DeadAGI on a live channel will cause problems, please use AGI -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----...

.... This is a new port that I am setting up. The reason why the switch went down was due to the amount of broadcast traffic coming from this port even though it was shutdown: ..snip.. TenGigabitEthernet7/4 is down, line protocol is down (notconnect) Hardware is C6k 10000Mb 802.3, address is 001d.4577.c693 (bia 001d.4577.c693) MTU 1500 bytes, BW 10000000 Kbit, DLY 10 usec, reliability 255/255, txload 1/255, rxload 1/255 Encapsulation ARPA, loopback not set Keepalive set (10 sec) Full-duplex, 10Gb/s input flow-control is off, output flow-control is off ARP type: ARPA, ARP Timeout...