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2007 Nov 06
4
MeetMe CPU resources
Hello, We would like to have a conference with 15 users aprox. We think that Pentium 4 3GHz and 1GB of RAM should be enough. Only Asterisk running. We wonder if somebody has some other experience, good or bad. We will use Asterisk 1.2 (it is a small and short project for only this). Thanks! -- Carles Pina i Estany GPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona
2009 Sep 16
3
Music on Hold
...#47;link2voip-sw1-02477668 -- Stopped music on hold on SIP/link2voip-sw1-02477668 No errors are printed, however the other side just hears silence. Here is the full debug output (asterisk -rvvvvv): == Using SIP RTP CoS mark 5 -- Executing [xxxxxxx at phones:1] Goto("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "1xxxxxxxxxx,1") in new stack -- Goto (phones,1xxxxxxxxxx,1) -- Executing [1xxxxxxxxxx at phones:1] MSet("SIP/ATA-xxxxxxxxxx-L1-024b6d88", "oldcidnum=0") in new stack -- Executing [1xxxxxxxxxx at phones:2] MSet("SIP/ATA-xxxxx...
2010 Apr 08
1
reshape panel data
...ing data set. The observation range over three years 1990-1992. But some cities did not report in some years. The original data looks like this: Cicoid year other_variables seclection-variable 1 1990 x x x x x x x 1 1 1991 xxxxxxxxxx 1 2 1991 xxxxxxxxxx 1 3 1990 xxxxxxxxxx 1 3 1991 xxxxxxxxxx 1 3 1992 xxxxxxxxxx 1 I would like to get a data s...
2006 Mar 30
0
SIP: INFO before answer causes disconnect
...P debug log (with some console verbosity as well) follows. Thank you very much for your help. Alan Ferrency pair Networks, Inc. alan@pair.com ------------------- SIP DEBUG LOG --------------------- -- Executing NoOp("Zap/3-1", "Dialing staff extension sip/sip_alan by XXXXXXXXXX") in new stack -- Executing Dial("Zap/3-1", "sip/sip_alan|20|o") in new stack We're at 10.11.220.1 port 13296 Adding codec 0x4 (ulaw) to SDP 13 headers, 8 lines Reliably Transmitting (no NAT) to 10.11.220.128:5060: INVITE sip:sip_alan@alan.phone SIP/2.0...
2006 Apr 10
2
Outbound calls through Broadvoice
...sulted the following two online tutorials: http://www.broadvoice.com/support_install_asterisk.html http://www.voip-info.org/wiki/view/Asterisk+settings+Broadvoice in an effort to make outbound calls. My current settings are as follows: sip.conf register => XXXXXXXXXX@sip.broadvoice.com:<SECRET>:XXXXXXXXXX@sip.broadvoice.com/XXXXXXXXXX where XXXXXXXXXX = our phone number including area code and <secret> is our broadvoice defined secret [sip.braodvoice.com] type=peer dynamic=yes username=XXXXXXXXXX fromuser=XXXXXXXXXX authname=XXXXXXXX...
2015 Feb 10
1
Dial Plan Issue
...nd and replays it. The FreePbx box works fine but the Standard Asterisk build is dropping the call during the first Voicemail playback and it does not leave the voicemail. Here is the printout of the log file for both boxes. Free PBX: [2015-02-10 12:12:34] VERBOSE[10502] pbx.c: -- Executing [XXXXXXXXXX at subMachine:4] Playback("SIP/trunk503out-00009728", "temp/0250002") in new stack [2015-02-10 12:13:29] VERBOSE[10502] pbx.c: -- Executing [XXXXXXXXXX @subMachine:5] Wait("SIP/trunk503out-00009728", "1") in new stack [2015-02-10 12:13:30] VER...
2006 Feb 06
1
Will not authenticate incoming VOIP provider calls
I running Asterisk 1.1 on Mandriva 2006. Everything works fine, can connect with softphone, send outgoing calls to VOIP provider. The only (and big) problem is that Asterisk refuses to authenticate incoming calls with the message (in the log): Failed to authenticate user "XXXXXXXXXX" <sip:XXXXXXXXXX@209.17.160.129> From what I've read in the various docs I could access, I should put insecure=very or insecure=port,invite (depending on the doc). I tried that and a lot of other things, nothing works. That message keeps coming back on every incoming calls. Her...
2017 Nov 16
2
Plugin virtual, Horde BAD IMAP QRESYNC not enabled
Return-path: <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx> Envelope-to: xxxxx at xxxxxxxxx Delivery-date: xxx, xx xxx xxxx xx:xx:xx +xxxx Received: xxxx [xxx.x.x.x] (xxxx=xxxxxxxxx) xx xxxxxxxxx.xxxxxxxxxxxx.xx xxxx xxxxx (xxxx x.xx) (xxxxxxxx-xxxx <xxxxxx-xxxxxxxx-xxxxxxxxx-xxxxxxx-xxxxxxx-xxx at xxxxxx.xxxxxxxxx.xx.xxx>) xx xxxxxx-xxxxxx-xx xxx xxxxx at xxxxxxxxx; xxx, xx xxx xxxx xx:xx:xx +xxxx Delivered-To: xxxxxxx.xxxxxxx at xxxxx.xxx Received: xxxx xxxxx-xxxx.x.xxxxxx.xxx [xx.xxx.xxx.x...
2005 Aug 21
2
Broadvoice Issue
..., so if it's there, I apologize for asking again. Starting about noon yesterday, I am no longer able to send/receive calls via Broadvoice. When calling in, I get a fast busy, and when calling out I get the following error: -- Executing Dial("SIP/112-572a", "SIPXXXXXXXXXX@sip.broadvoice.com") in new stack -- Called XXXXXXXXXX@sip.broadvoice.com Aug 21 13:34:47 NOTICE[20742]: chan_sip.c:8648 handle_response: Failed to authenticate on INVITE to '"Mobile" <sip:XXXXXXXXXX@sip.broadvoice.com>;tag=as124e3440' == Spawn extension (agent...
2013 Feb 26
1
set time zone in sip debug logs
Hello, Please suggest the way to change the time zone in below sip debug logs. INVITE sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060 SIP/2.0Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7bbd9;rportMax-Forwards: 70From: "xxxxxxxxxx" <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx>;tag=as23a29r59To: <sip:xxxxxxxxxx at xxx.xxx.xxx.xxx:5060>Contact: <sip:xxxxxxxxxx at xxx.xxx.xxx...
2005 Mar 09
2
Broadvoice latest changes and still not working-An
...is message when I reload my configs files. Mar 9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response: Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec (Scheduling reregistration in 1933000 ms) Below is the debug: -- Executing Dial("OSS/dsp", "SIP/xxxxxxxxxx@sip.broadvoice.com|30") in new stack We're at outsideIPaddress port 14842 Answering with preferred capability 0x4 (ulaw) 12 headers, 8 lines Reliably Transmitting: INVITE sip:xxxxxxxxxx@proxy.lax.broadvoice.com SIP/2.0 Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8...
2009 Aug 21
1
Queue Question
...m/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten => XXXXXXXXXX,1,Verbose(2,Logging in agent) exten => XXXXXXXXXX,n,WaitExten(5) exten => XXXXXXXXXX,n,GoSub(AgentCallbackLogin,start,1) exten => XXXXXXXXXX,n,Hangup() [AgentLogOut] exten => XXXXXXXXXX,1,RemoveQueueMember(9819930,DAHDI/g1/${CALLERID(num)}) ; calling 'pri...
2005 Mar 08
1
All Circuits are Busy Now
...; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown #include sip_nat.conf #include sip_additional.conf register => xxxxxxxxxx@sip.broadvoice.com:pppppppppp:xxxxxxxxxx@sip.broadvoice.com/2197 [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=xxxxxxxxxx secret=pppppppppp username=xxxxxxxxxx insecure=very context=from-broadvoice authname=xxxxxxxxxx dtmfmode=inband d...
2005 Mar 09
4
Broadvoice Multiple "lines"
I configured this once now I forgot what I did. Two Broadvoice accounts. Incoming is simple - just use the phone numbers. Outgoing: Dial out on a specific line and/or set up the groups and select the other "line" if the first one is busy? -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956
2010 Feb 02
2
Semi-Transfer
...> X,5,Monitor(wav,/var/store/calls/InOutRec-${STRFTIME(${EPOCH},,%Y%m%d-%H %M%S)}-${CALLERID(num)}-${EXTEN},mb) exten=> X,6,dial(${belltd}/${num}) Here is what I see in the CMD when the dial fails -- Timeout on DAHDI/52-1, continuing... -- Executing [XXXXXXXXXX at RecOut:5] Monitor("DAHDI/52-1", "wav,/var/store/calls/InOutRec-20100202-133012-XXXXXXXXXX-XXXXXXXXXX,mb" ) in new stack -- Executing [XXXXXXXXXX @RecOut:6] Dial("DAHDI/52-1", "DAHDI/G3/4099819750") in new stack --...
2012 Jan 13
0
Samba mixing Domain & Server name
...er name of Browse Master ------------- ---------------------------- TASC TUX #pdbedit -Lvu "user" gives Unix username: user NT username: Account Flags: [U ] User SID: S-1-5-21-64526847-XXXXXXXXXX-XXXXXXXXXX-1002 Primary Group SID: S-1-5-21-64526847-XXXXXXXXXX-XXXXXXXXXX-513 Full Name: Home Directory: HomeDir Drive: H: Logon Script: logon.bat Profile Path: Domain: TUX Account desc: Workstations: Munged dial: Logon time: 0 Logoff time:...
2006 Nov 29
12
What's up with the Manager Interface?!?!
The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.
2014 Feb 23
1
Problem with cron
...job that executed correctly at 6pm was executing minutes past 6pm when the server was restarted. This is totally unexpected behavior. Here is the output of the cron log file. The name of the server has been redacted. # grep -i 'Feb 22' cron-20140223 | grep -i poweroff Feb 22 18:00:01 xxxxxxxxxx CROND[2875]: (root) CMD (poweroff) Feb 22 18:12:01 xxxxxxxxxx CROND[1894]: (root) CMD (poweroff) Feb 22 18:16:01 xxxxxxxxxx CROND[1893]: (root) CMD (poweroff) Feb 22 18:18:01 xxxxxxxxxx CROND[1896]: (root) CMD (poweroff) Feb 22 18:22:01 xxxxxxxxxx CROND[1915]: (root) CMD (poweroff) Feb 22 18:25:01...
2020 Jun 12
0
Forbidden call
...alaw to SDP Adding codec gsm to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (NAT) to X.X.X.X :1024: INVITE sip:2012@ X.X.X.X :1024;ob SIP/2.0 Via: SIP/2.0/UDP X.X.X.X :5060;branch=z9hG4bK2555a6ef;rport Max-Forwards: 70 From: "Jerry Geis 101" <sip:XXXXXXXXXX@ X.X.X.X >;tag=as5e61ec66 To: <sip:2012@ X.X.X.X :1024;ob> Contact: <sip:XXXXXXXXXX@ X.X.X.X :5060> Call-ID: 361b4b803f214946320c0af84a9ac0c4@ X.X.X.X :5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 13.33.0 Date: Fri, 12 Jun 2020 12:18:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE,...
2005 May 10
0
outbound PSTN numbers over SIP failing
...VERBOSE[1563]: -- Executing [1;36;40mSetCIDNum[0;37;40m("[1;35;40mSIP/200-8f7f[0;37;40m", "[1;35;40m31437110323[0;37;40m") in new stack May 8 10:47:11 VERBOSE[1563]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/200-8f7f[0;37;40m", "[1;35;40mSIP/XXXXXXXXXX@budgetphone.nl|30|r[0;37;40m") in new stack May 8 10:47:11 DEBUG[1563]: SIMPLE DIAL (NO URL) May 8 10:47:11 DEBUG[1563]: Outgoing Call for XXXXXXXXXX May 8 10:47:11 DEBUG[1563]: XXXXXXXXXX is not a local user May 8 10:47:11 VERBOSE[1563]: -- Called XXXXXXXXXX@budgetphone.nl May 8 10:...