search for: 6001

...55 aadk*CLI> Core debug was 0 and is now 255 [Kaadk*CLI> core set verbose 255 aadk*CLI> Verbosity was 0 and is now 255 [Kaadk*CLI> <--- SIP read from 192.168.220.31:5060 ---> REGISTER sip:asterisk.foo.internal SIP/2.0 Call-ID: 2000fb00-4bca-c0a8efe3 at asterisk.foo.internal From: 6001<sip:6001 at asterisk.foo.internal>;tag=10007c00-4bc9 To: 6001<sip:6001 at asterisk.foo.internal> CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.220.31:5060;branch=z9hG4bKc0a8efe310007f004bce Contact: sip:6001 at 192.168.220.31:5060 Max-Forwards: 70 User-Agent: LRSTD LR8882 2.5.00_99 Expire...

...ave upgraded my server to RedHat EL4, and all my Windows users when the username have Uppercases in username can't acces to my samba shares, and all other users works. I modify my log level to 10 and when any user try access to samba server the log message is: May 3 11:08:48 wichita smbd[6001]: [2005/05/03 11:08:48, 0] auth/auth_util.c:make_server_info_info3(1134) May 3 11:08:48 wichita smbd[6001]: make_server_info_info3: pdb_init_sam failed! May 3 11:08:48 wichita smbd[6001]: [2005/05/03 11:08:48, 0] auth/auth_util.c:make_server_info_info3(1134) May 3 11:08:48 wichita smbd[6001...

Happened twice. There might also be a race condition and some bad pointers in chan_local.locals_show. First the segfault. CLI> show locals <unowned> -- 6001@default Segmentation fault (core dumped) [root@mars asterisk]# ll -tr total 22260 [...] Loaded symbols for /usr/lib/asterisk/modules/chan_local.so #0 __pthread_mutex_lock (mutex=0x5d8) at mutex.c:99 99 mutex.c: No such file or directory. in mutex.c (gdb) bt #0 __pthread_mutex_lock (...

...ity as they're the cheapest by far...). 3. Tried recompiling asterisk from source, just in case the debian package was broken. I still get the error: May 17 23:20:27 NOTICE[1110916016]: chan_sip.c:5059 handle_response: Failed to authenticate on INVITE to '"Tony Hoyle" <sip:6001@213.208.99.114>;tag=as5c348356' Relevant chunks here of data are: [pipecall] type=peer secret=xxxx username=xxxx host=sipproxy.pipecall.com [6001] type=friend username=6001 secret=xxxx host=dynamic context=inbound-from-local The log looks like: Sip read: INVITE sip:8378@asterisk SIP/2.0...

Hi. I my dialplan I have : same = n,Dial(PJSIP/6001,10) same = n,Dial(PJSIP/6002,30) same = n,Hangup() The extension 6002 will not be invited until the called party 6001 hangs up or until 10 seconds if nobody answers the call in 6001. How to call 6001 and immediately call 6002, having 2 phones ringing at same time, but without doing something li...

Hi SamyGo. Thank you for the replay. So, let me explain it better: I knew that I could use something like " same = n,Dial(PJSIP/6001&PJSIP/6002) ". While every extension (called phones) rings and before anyone answers, SIP 183 messages will be sent to Asterisk from callees. If a called phone answer, the others will be hanged up. It is ok for me. I want to connect the caller just to the first called party that answers....

...12 ISeqno: 009 Type: IAX Subclass: ACK Timestamp: 27464ms SCall: 00002 DCall: 08434 [127.0.0.1:40310] [Jul 7 14:41:11] DEBUG[1261]: chan_iax2.c:8273 socket_process: For call=2, set last=27464 [Jul 7 14:41:11] DTMF[9852]: channel.c:2400 __ast_read: DTMF end '*' received on IAX2/6001-2, duration 0 ms [Jul 7 14:41:11] DTMF[9852]: channel.c:2436 __ast_read: DTMF begin emulation of '*' with duration 100 queued on IAX2/6001-2 [Jul 7 14:41:11] DEBUG[9852]: channel.c:4163 ast_generic_bridge: Got DTMF begin on channel (IAX2/6001-2) [Jul 7 14:41:11] DEBUG[9852]: channel.c:...

...net=10.50.55.0/24 external_media_address=<scrubbed public ip> external_signaling_address=<scrubbed public ip> cert_file=/etc/asterisk/keys/dev1.crt priv_key_file=/etc/asterisk/keys/dev1.key ca_list_file=/etc/asterisk/keys/ca.crt cipher=AES256-SHA method=tlsv1 ;===============EXTENSION 6001 [6000] type=endpoint context=internal disallow=all allow=ulaw auth=auth6000 aors=6000 direct_media=no rewrite_contact=yes ; necessary if endpoint does not know/register public ip:port ice_support=no force_rport=yes rtp_symmetric=no media_encryption=sdes [auth6000] type=auth auth_type=userpass...

...ho de 2015 17:43 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [asterisk-users] RES: How to dial extensions asynchronous-sequentially ? All I can focus now is "the objective is to see if there is an way to deliver more than one SIP 183 message to the caller" 6001 has a song playing in 183 and 6002 has a "service unavailable" message, do you intend to deliver both of them simultaneously to the caller? I've seen multiple 183 Session Progress messages getting delivered to caller but what is your end game ? Play all sort of messages to the caller...

Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

Hello. I am having a lot of problems with SIP through NAT. So, I decided to adopt PJSIP. However, I am not able to make the extensions ring when receiving a call from the queue. I'm using telnet to include the extension and on the asterisk console, it even shows Called PJSIP/6001, but the extension doesn't ring. If I call from extension to extension, it works normally. telenet: Action: QueueAdd Queue: queuetest MemberName: 1234 Interface: PJSIP/6001 StateInterface: PJSIP/6001 Ringinuse: yes Paused: false If I change to SIP, the extension will call normally. My conf...

Hello, I installed 1.6.2.17 version of asterisk. Set the user database to realtime. I have no problems with sip users. They can register talk etc.. With iax clients, they can register also.. And when they call iax to sip, it works. When they make an echo test..no voice received on iax clients. When they make call from sip to iax ..no sound received on iax clients. I didnt see any clue on debug.

...it is an issue with Asterisk. sip.conf: > ;SIP-Phones (Twinkle) > [user1] > callerid = 6000 > username = 6000 > secret = 6000 > canreinvite = no > type = friend > context = phones > allow = all > host = dynamic > dtmfmode = info > > [user2] > callerid = 6001 > username = 6001 > secret = 6001 > canreinvite = no > type = friend > context = phones > allow = all > host = dynamic > dtmfmode = info > > ; Mobile phone > [123456789101112] > callerid = 6201 > username = 6201 > secret = 6201 > canreinvite = no > ty...

Hello, I made an interesting observation related to the "Clocksource tsc unstable (delta = -2999660320319 ns)" problem. In the log of ntpd I found: Oct 5 03:46:35 greenville-dom0 ntpd[4020]: kernel time sync status change 6001 Oct 5 04:03:41 greenville-dom0 ntpd[4020]: kernel time sync status change 2001 Oct 5 05:29:03 greenville-dom0 ntpd[4020]: kernel time sync status change 6001 Oct 5 05:46:09 greenville-dom0 ntpd[4020]: kernel time sync status change 2001 Oct 5 06:54:30 greenville-dom0 ntpd[4020]: synchronized...

...responds an ACK and marks the peer as REACHABLE - and now history repeats itself indefinitely, switching between UNREACHABLE and REACHABLE after each POKE. This results in these notices in the logs: [Aug 21 10:54:58] NOTICE[13318] chan_iax2.c: Peer 'remote-host' is now REACHABLE! Time: 6001 [Aug 21 10:56:02] NOTICE[13323] chan_iax2.c: Peer 'remote-host' is now UNREACHABLE! Time: 6001 [Aug 21 10:56:38] NOTICE[13319] chan_iax2.c: Peer 'remote-host' is now REACHABLE! Time: 6001 [Aug 21 10:57:42] NOTICE[13324] chan_iax2.c: Peer 'remote-host' is now UNREACHABLE!...

...hannel is still available on it. I would like to be able to differentiate between these two cases: no calls engages, or calls engaged. Is there any way to do that? Turning off call waiting on the snom200 is not an option. Here's why I need this to work: exten => 9,2,ChanIsAvail(SIP/6001) exten => 9,3,Dial(SIP/6001,20) exten => 9,103,Dial(SIP/6001&SIP/6002&SIP/6003) So if 6001 isn't already on the phone, only their phone rings. If 6001 is on the phone, then the call is sent to 2 other phones as well 6001. Can this be done? Is there a better way? Thanks very m...

The Wiky needs to be updated https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip#Configuringres_pjsip-IDENTIFY%28res_pjsip_endpoint_identifier_ip%29 This is the example shown: "[6001] endpoint=6001 match=203.0.113.1" It should be: "[6001] type=identify endpoint=6001 match=203.0.113.1"

...l/9991416445 at from-internal-0000017b;2 -- SIP/5547741200-000051cb is ringing -- SIP/5547741200-000051cb is making progress passing it to Local/9991416445 at from-internal-0000017b;2 -- SIP/5547741200-000051cb answered Local/9991416445 at from-internal-0000017b;2 -- Executing [6001 at from-internal:1] Macro("Local/9991416445 at from-internal-0000017b;1", "user-callerid,") in new stack -- Executing [s at macro-user-callerid:1] Set("Local/9991416445 at from-internal-0000017b;1", "TOUCH_MONITOR=1429224932.21696") in new stack --...

...;/usr/local/etc/rc.d/pound.sh start" stop program = "/usr/local/etc/rc.d/pound.sh stop" if totalmem > 400.0 MB for 5 cycles then restart if failed port 6000 protocol http with timeout 10 seconds then restart group server #check mongrel processes #6001 check process mongrel-6001 with pidfile /home/xxx/sshare/app/log/mongrel.6001.pid start program = "/home/xxx/scripts/mongrel_rails_start 6001" stop program = "/home/xxx/scripts/mongrel_rails_stop 6001" if totalmem > 50.0 MB for 5 cycles then restart if failed...

...I've posted all the config that I felt was relevant here, let me know if you need more. This was all written by Asterisk-GUI. I realise there's a lot more configuration but given that things work fine when I set the receive to a single agent, I assumed it was a queue issue. Users.conf [6001] callwaiting = yes context = numberplan-custom-1 email = jsmall at visinet.com.au fullname = Joshua Small hasagent = yes hasdirectory = yes hasiax = no hasmanager = no hassip = yes hasvoicemail = no host = dynamic mailbox = 6001 secret = SECRET threewaycalling = yes registeriax = no...