search for: 401

...gent,all Queue: 140 Location: Local/402 at default/n <mailto:%20Local/402 at default/n> Paused: 1 Event: QueueMemberPaused Privilege: agent,all Queue: 140 Location: Local/402 at default/n <mailto:%20Local/402 at default/n> Paused: 0 Event: Newchannel Privilege: call,all Channel: SIP/401-08197170 State: Ring CallerID: 401 CallerIDName: JOSEP Uniqueid: 1196854142.4 Event: Newexten Privilege: call,all Channel: SIP/401-08197170 Context: default Extension: 140 Priority: 1 Application: Answer AppData: Uniqueid: 1196854142.4 Event: Newstate Privilege: call,all Channel: SIP/401-0819717...

...virEventPollUpdateTimeout:265 : EVENT_POLL_UPDATE_TIMEOUT: timer=2 frequency=5000 2018-11-13 14:21:39.942+0000: 95205: debug : virEventPollUpdateTimeout:282 : Set timer freq=5000 expires=1542118904942 2018-11-13 14:21:39.942+0000: 95205: debug : virEventPollInterruptLocked:722 : Skip interrupt, 1 140197681715264 2018-11-13 14:21:39.942+0000: 95205: debug : virNetServerClientSendMessageLocked:1480 : msg=0x561dca9e2880 proc=1 len=28 offset=0 2018-11-13 14:21:39.942+0000: 95205: info : virNetServerClientSendMessageLocked:1488 : RPC_SERVER_CLIENT_MSG_TX_QUEUE: client=0x561dca9e6e60 len=28 prog=18018...

Hi, I have made a fresh install of asterisk-1.6.2.10 and when I register my soft phone it gives following error. Rest are default configurations. 32.454370 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER sip:ASTERISK_SERVER_IP 32.454505 67.19.43.202 -> MY_IP SIP Status: 401 Unauthorized (0 bindings) 36.454814 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER sip:ASTERISK_SERVER_IP 36.454949 67.19.43.202 -> MY_IP SIP Status: 401 Unauthorized (0 bindings) 40.467996 MY_IP -> ASTERISK_SERVER_IP SIP Request: REGISTER sip:ASTERISK_SERVER_IP 40.468132 67.1...

...tion(runs,seed){ .Random.seed<-seed for (i in 1:runs) { print(i) print(runif(1,0,1)) print(.Random.seed)} return(.Random.seed} Consider the following input/output: >RNGkind(kind="Marsaglia-Multicarry") > set.seed(20391) > seed1<-.Random.seed > seed1 [1] 401 -1607331462 -462081869 > test(2,seed1) [1] 1 [1] 0.4188851 [1] 401 -1607331462 -462081869 [1] 2 [1] 0.7713649 [1] 401 -1607331462 -462081869 [1] 401 -1607331462 -462081869 > seed1 [1] 401 -1607331462 -462081869 > test(2,seed1) [1] 1 [1] 0.7293294 [1]...

Through the AMI how can I tell if a call is on hold or not? I am using 1.4.X Thanks, jerry

...hosted by Apache using Digest authentication. It migrated from Apache 2.2.9 (Debian 5) to Apache 2.2.15 (CentOS 6). The internal hostname for the wiki server is a DNS CNAME that was repointed from one host to another during the cutover. In a normal session, 1. Client sends GET 2. Server sends 401 with WWW-Authenticate header 3. Client sends GET with Authorization header 4. Server sends 200 with Authentication-Info header and page content When using IE 9 (and possibly 8, I'm still trying to track that down), 1. Client sends GET 2. Server sends 401 with WWW-Authenticate header 3. Client...

...est Bad or missing password on admin command request (command: stats.xml) [2019-01-01 13:10:32] INFO admin/admin_handle_request Bad or missing password on admin command request (command: stats.xml) access.log 192.168.0.7 - - [01/Jan/2019:13:08:31 +0000] "GET /admin/stats.xml HTTP/1.1" 401 360 "-" "sambc/2018.10" 0 192.168.0.7 - - [01/Jan/2019:13:08:31 +0000] "GET /admin/stats.xml HTTP/1.1" 401 360 "-" "sambc/2018.10" 0 192.168.0.7 - - [01/Jan/2019:13:08:31 +0000] "GET /admin/stats.xml HTTP/1.1" 401 360 "-" "s...

...à 11:54, Administrator a écrit : > > > > Le 15/01/2020 à 19:24, Administrator a écrit : > >> Hi all, > >> > >> we face a strange behavior while connecting an Asterisk16 instance > >> with PJSIP to 2 providers: we receive error 401 Unauthorized, > both of > >> them having Kamailio as front-end. With other providers -we don't > >> know if they run kamailio- registration is just fine. > >> > >> One of the provider took a pcap and told us that expiration was >...

Hi! I would like to divide a 401 by 401 image into 20 equal blocks and then re-plot the image with the segments randomly distributed in it. How can I do this? ............. Regards, Kenduiywo Benson [[alternative HTML version deleted]]

I have a SIP trunk - calls going out work fine. Trying to setup an incoming call with a DNIS When I dial the number - I see nothing on the CLI. The person says the server is returning 401 How do I debug that. Using asterisk 18.8.0 Thanks Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20230309/4c5d5979/attachment.html>

...and issue I hope someone can help with.. I have calls that often come in that need to be blocked. We wish to do this without answering the call. The issue is our carriers have fail over servers and will try sending the call from each when we block the call. If we send a hangup with a Sip 401 they will stop the route advance on their end. The issues is we have been sending a hangup/cause code 21 (Call rejected) But they are receiving a 403 Forbidden.. Is there any hangup code that we can send that will reply a 401.. We see the 401 on the inbound is converted to a cause code...

Le 17/01/2020 à 11:54, Administrator a écrit : > > Le 15/01/2020 à 19:24, Administrator a écrit : >> Hi all, >> >> we face a strange behavior while connecting an Asterisk16 instance >> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of >> them having Kamailio as front-end. With other providers -we don't >> know if they run kamailio- registration is just fine. >> >> One of the provider took a pcap and told us that expiration was set >> to 0 that's why they don't ac...

I am getting a SIP 401 unauthorized message. My public IP or PIP is being pre-routed with iptables to goto an internal IP or IIP All the polycom phones in the office point to the IIP. they work fine. I have 2 external phones that are registering to the PIP. I see the register attempt as I am getting the 401 unauthoriz...

I have 2 ip phones linksys spa921 and 1 normal phone connected to a cisco spa8800, all them are internal lines. 1.- spa921, 401 ext 2.- spa921, 402 ext 3.- normal phone connected to spa8800 404 ext. It had a very strange behavior when I was configuring call transfer and call pickup. These are steps to repeat it: 1.- from 401 call to 404 2.- from 404 don't answer it. 3.- from 402 press *8 and wait 10 seconds 4.- 402...

...(listmounts.xsl) [2004-02-14 09:40:23] DBUG admin/command_list_mounts List mounts request [2004-02-14 09:40:23] DBUG admin/admin_send_response Sending XSLT (/share/icecast/admin/listmounts.xsl) My access.log contains 192.168.1.21 - - [14/Feb/2004:09:26:57 +0000] "SOURCE / ICY/666" 401 115 "-" "-" 0 192.168.1.21 - - [14/Feb/2004:09:31:06 +0000] "GET /admin.cgi HTTP/1.0" 404 109 "-" "liveice-xmms" 0 192.168.1.21 - - [14/Feb/2004:09:31:06 +0000] "SOURCE / ICY/666" 401 115 "-" "-" 0 192.168.1.21 - - [14/...

Hi ? I am trying to get 2 accounts from voipblaster to talk to each other. Calls withing voipblaster network is free. If I configure two sip clients?with the two accounts it works fine however with Asterisk I am getting SIP 401 ? In my Sip.conf file I?under general ? register = user:password at sip.voipblaster.com ? then I have a sip peer ? ? [FreeCall](default) type= friend context= incoming username = kiks2010 secret = password host= sip.voipblast.com fromuser = kiks2010 fromdomain = sip.voipblast.com insecure=very qual...

...:42 PM Jerry Geis <jerry.geis at gmail.com> wrote: > >> I have a SIP trunk - calls going out work fine. >> >> Trying to setup an incoming call with a DNIS >> >> When I dial the number - I see nothing on the CLI. >> The person says the server is returning 401 >> >> How do I debug that. Using asterisk 18.8.0 >> >> Thanks >> >> Jerry >> > > Thanks I am using chan_sip. Turning on "sip set debug on" I do se it. > > > > Using INVITE request as basis request - > 0gQAAC8WAAACBAAALxYAAJJTf...

...lt;admin at tootai.net> wrote: <snip> > It become stranger and stranger: on one of the register peer we receive in > asterisk: > > *CLI> [2020-01-19 15:23:18] WARNING[17469]: > res_pjsip_outbound_registration.c:1021 handle_registration_response: Fatal > response '401' received from 'sip:<myprovider>' on registration attempt to > 'sip:<myuser>@<myprovider>', stopping outbound registration > What is the actual full configuration for this? > On the other one: > > [2020-01-19 15:23:46] WARNING[17469]: > res_p...

http://bugzilla.mindrot.org/show_bug.cgi?id=401 ------- Additional Comments From yoshfuji at linux-ipv6.org 2003-01-06 12:20 ------- Created an attachment (id=194) --> (http://bugzilla.mindrot.org/attachment.cgi?id=194&action=view) patch to run openssh-3.5p1 on linux-2.2 & glibc-2.2(or later) This is modified patch for openssh...

...d i'm setting up a asterisk box to register around 100 lines from my central softswitch. This asterisk box will be placed inside a customer and has a digium card to be interconected with customer's pabx. My problem is that when asterisk send register message, my softswitch return with sip 401 and asterisk should send a register message with Authorization in header. Only after 10, or 15 or 30 or 43 or 2 messages 401 asterisk start to send Authorization in header. This is a random time, don't follow any rule. This problem cause lines disregistration some times during a day. How can...