similar to: Members in 'Unknown' status in output of 'queue show'

Hi, I am using asterisk-1.4.15, My sip configs is like [2501] type=friend username=2501 secret=2501 canreinvite=no host=dynamic dtmfmode=rfc2833 context = sip disallow=all allow=ulaw incominglimit=1 nat=1 queue.conf is like [gen-enq] joinempty = yes musiconhold = default strategy = rrmemory servicelevel = 60 timeout = 60 retry = 5 wrapuptime=5 announce-frequency = 90 announce-holdtime = yes

Hi, I have an Asterisk 1.2.9.1 box on a Debian distro with mISDN drivers. I installed the new driver (0.3.1-rc30) on our pbx but since no voice was passing I decided to go back to old version (0.3.1-rc23). Last friday everything seemed to work fine but now every incoming call drops after 3-4 seconds while Asterisk console is showing these messages: Apr 23 12:42:39 DEBUG[7625]:

I attempted an upgrade of our production system from Asterisk/Zaptel 1.2 to 1.4 this weekend. Intially everything looked like it was working properly, but some time in the day following the upgrade, the system died to a kernel panic. I wasn't able to catch the entire kernel dump on the console unfortunately. I attempted to isolate the panic, and found that when 'service zaptel stop'

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

https://bugzilla.mindrot.org/show_bug.cgi?id=1410 Summary: Correct UsePAM comment in sshd_config on Mac OS X Classification: Unclassified Product: Portable OpenSSH Version: 4.7p1 Platform: Other OS/Version: Mac OS X Status: NEW Severity: normal Priority: P2 Component: PAM support AssignedTo:

Hi R-users, I have missing data for the month. My question is how do I insert the missing month and fill up the cell with 'na' for the rain amount?  Then I would like to count the percentage of missing data. No     Year     month rain 1398 1985    10 104.2 1399 1985    11 138.0 1400 1985    12 120.4 1401 1986     1  12.6 1402 1986     2  19.4 1403 1986     3   1.0 1404 1986     4  58.8

Hi, calling a partner on the other side of a SIP trunk, call gets disconnected immediately after answer. This is the content of log file: Jun 14 16:25:14 DEBUG[14380] channel.c: Didn't get a frame from channel: SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Bridge stops bridging channels SIP/232-2e41 and SIP/cerved-out-6eba Jun 14 16:25:14 DEBUG[14380] channel.c: Hanging up

Hi, Anyone can tell me what this means? [Feb 3 12:42:32] WARNING[12130]: chan_sip.c:3293 update_call_counter: Inringing for peer 'test-peer' < 0? Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090203/a39503b7/attachment.htm

Hi, I am trying to configure busy detect on sip channel but somehow its not working may be this is my mistake could you please help me to figure out. I have added following options in my sip.conf [7527] type=friend context=from-sip host=dynamic dtmfmode=rfc2833 callerid="Guest" <7527> mailbox=7527 at default nat=no qualify=yes cc_agent_policy=generic cc_monitor_policy=generic

Hi all. We have a problem with correct dialstatus and cdr(disposition) when using call-limit. When call-limit reached dialstatus is CHANUNAVAIL and CDR(disposition)='NO ANSWER' -- Executing [0014 at sub_pbxdialco:49] Dial("SIP/1295-000001f8", "SIP/0014,12,tTkK") in new stack == Using SIP RTP CoS mark 5 [2013-07-03 15:22:27] NOTICE[29728]: chan_sip.c:6003

Hi, I have declared my sip users call-limit=2 and type=friend. When any user recieves a waiting call while already in a conversation, the peer call counter is set to 2.The problem is that, the counter is not reset to zero after hangup and becoz of this the user is not able to recieve any call anymore even if s/he has hungup. the asterisk cli displays the following error. [Mar 17 16:15:10]

I just tried to go from CAS to PRI on my T1 (Sangoma), and failed pretty badly. Seemingly everything worked -- Asterisk would see the incoming call (including CID and DID info), try to route it, and fail -- giving me a telco (not Asterisk) call failure message. My zapata.conf and zaptel.conf files are at http://pastebin.com/558349 Below's the log dump. Note that, because I was simply going

Hi, Does any one experience that SIP phone to SIP phone (Polycom phone) calls can't hear each other, but Monitor application records both end's voices. It also happens in group pickup calls. Zap calls to queue (Local channel) also experience this problem (sometimes, our SIP phone can't hear any voice from incoming Zap calls when pickup, sometimes this happens after 10-50

Hello, Can somebody please assist me with my problem. Currently I am using a Asterisk@HOme version 2.4 with a TE406P digium card. One the E1 is connected to a telco switch via an ISDN. May issue is that may incoming calls in the zap channels gets disconnected or release after 1 ring. I really dont know what setting should I change to increase the timeout of the ring. I have even tried upgrading

Hi, My asterisk sometimes stop responding to iax calls. In the log, I've found this: Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) - decrement call limit counter Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29

I tried to add a couple of SIP phones (polycom 601s) to my existing asterisk installation. I can successfully make a call from the SIP phone to any other phone (inside or outside), but I can not make any calls to a SIP phone. Attached are the pertinent parts of sip.conf and extensions.conf. The log starts off normal with: Mar 26 09:51:15 DEBUG[4885] chan_zap.c: DTMF digit: 2 on Zap/55-1 Mar

I need some extensions logic assistance, I'm trying to dial out one of multiple SIP trunks, in sequence. I need to detect a busy SIP trunk(I only allow 1 call per trunk) and roll over to a second or third depending on that busy status Here's what I've got for a macro thusfar, but it's not working(fails if the 1st trunk is busy) extensions.conf: [globals] trunk_1 => SIP/trunk1

In syslog of my DC (2:4.5.12+dfsg-2+deb9u2~bpo8+1) i found sometime rows like: Mar 21 09:53:40 vdcsv1 smbd[22686]: [2018/03/21 09:53:40.826081, 0] ../source3/param/loadparm.c:3244(process_usershare_file) Mar 21 09:53:40 vdcsv1 smbd[22686]: process_usershare_file: stat of /var/lib/samba/usershares/sysvo failed. Permesso negato Mar 21 09:53:40 vdcsv1 smbd[22686]: [2018/03/21 09:53:40.831949,

Guys. Ive seen on my asterisk messages log that asterisk has shutdown itself about 12 times in 5 days... The logs show nothing but: [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Drop call [Jun 28 09:40:02] WARNING[3172]: Unicall/4 event Release call [Jun 28 09:40:02] VERBOSE[3172]: [Jun 28 09:40:02] -- Unicall/4 released [Jun 28 09:40:02] VERBOSE[3084]: [Jun 28 09:40:02] Asterisk cleanly

Hi I have noticed that asterisk returns 'SIP 603' when the called party does not answer. My test setup is simple: two SIP phones (extensions: 100 and 111) registered to an Asterisk 1.8.30.0 gateway.The Dial timeout is 30 seconds. When 100 calls 111 and after 30 seconds, asterisk sends a CANCEL request to 111 (expected) and a '603 Decline' response to 100 (unexpected &