search for: 1234567890

Hi Folks, Compare print(1234567890,digits=4) # [1] 1.235e+09 print(1234567890,digits=5) # [1] 1234567890 Granted that digits: a non-null value for 'digits' specifies the minimum number of significant digits to be printed in values. how does R decide to switch from the "1.235e+09" (rounded to 4 digi...

...ver is not transmitted to the E1, the calling party hears no dial tone, but also no hangup or congestion or anything else just nothing. This is the console log of the E1 server passing the call via e164 resolution an SIP to another server: -- Executing EnumLookup("Zap/1-1", "1234567890") in new stack -- Accepting call from '' to '1234567890' on channel 0/1, span 1 -- Executing SetAccount("Zap/1-1", "0.00") in new stack -- Executing Dial("Zap/1-1", "SIP/1234567890@host.domain.tld|180|r") in new stack...

...erisk version 1.6.1.12. The specific requirements of the gateway in the configuration I am trying to use specify that the Name part of the From header be blank with the outbound number that needs to be dialed in the number field of the From header. So I want it to look like this: From: <sip:1234567890 at 192.168.3.110>;tag=xxx However, even if I set the name to blank, using Set(CALLERID(name)= ), Asterisk always seems to put the CallerID number in the name field as well and here is what I get: From: "1234567890" <sip:1234567890 at 192.168.3.110>;tag=xxx I cannot figure out...

Hello, I would like to verify if a specific SIP header exists, and if yes, extract the partial content from another header. 1. Is there a way to verify if a specific header exists? 2. How do I extract data that is between the first : and the following @? Specifically, The data looks like <sip:1234567890 at 10.0.0.1:5060> and I would like to get only the 1234567890 I tried to use the CUT() command, but without success. Thank you. Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20101005/bbd8350...

Thank you for the speedy reply. My originate string is something like the following where xxxxx is really the sip provider's supplied IP address 1234567890 is really the phone number I am dialing PJSIP/outbound.vitelity.net/1234567890 In the chan_sip based solution, it's... SIP/outbound.vitelity.net/1234567890 Have a great day! Dan -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at li...

Hi, Since we cannot join a W10 machine to NT3.x domain anymore, it is time to move on. We have a decade-old domain 'A1CWB' and will profit from the situation fixing the old S-1-5-21-1234567890-1234567890-1234567890 SID and implementing a new domain name: Old domain: A1CWB, SID S-1-5-21-1234567890-1234567890-1234567890 New domain: AD.A1.IND.BR, decent SID from net getdomainsid, two servers, one DC and one DM as fileserver, Ubuntu 18.04. On my tests I was able to import old LDAP ac...

...Ext: 1 Coding: 0 Number Specified Channel Type: 3 > Ext: 1 Channel: 23 ] > [1e 02 81 82] > Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) > I would like to print '1234567890 Message type: CONNECT (7) ....... ....... ' where 1234567890 is the callerid Thanks Regards -- Arpit Mehta Graduate Student Department of Computer Science Columbia University Tel: 1-646-387-5998

...ss added interface ip=192.168.2.3 bcast=192.168.2.255 nmask=255.255.255.0 querying pdc on 192.168.2.255 name_query failed to find name pdc bdc:/ # smbclient -L pdc success, all services, shares and workstations are listed pdc:/ # net getlocalsid SID for domain THEDOMAIN is: S-1-5-21-1234567890-1234567890-1234567890 bdc:/ # net getlocalsid [2006/11/08 12:53:58, 0] utils/net.c:net_getlocalsid(494) Can't fetch domain SID for name: BDC To complete the picture, running 'nbtstat -a bdc' and 'nbtstat -a pdc' works just fine from the local Windows client's DOS p...

...at the end of the dial string just seems to prevent the call from going through. Username is your 10-digit phone number. Password is in the .reg file they sent you via email. I signed up for the USB phone, so I don't if they send a .reg file if you went for the MTA. sip.conf register => 1234567890:password@myphone.voiceglo.com [myphone.voiceglo.com] type=peer username=1234567890 secret=password nat=no host=myphone.voiceglo.com disallow=all allow=ulaw allow=alaw canreinvite=no qualify=200 restrictid=no fromuser=1234567890 fromdomain=1234567890.voiceglo.com extensions.conf exten => _91NX...

...oesn't see the domain until I manually add an entry for it in /etc/passwd Checking the user entry for two users I can find the following differences. BERENICE is an user imported from the old system and is working fine: dn: uid=berenice,ou=Users,dc=DOMAIN,dc=IT uid: berenice sambaSID: S-1-5-21-1234567890-123456789-123456789-2018 sambaPrimaryGroupSID: S-1-5-21-1234567890-123456789-123456789-513 displayName: berenice sambaLogonTime: 0 sambaLogoffTime: 4294967295 sambaKickoffTime: 4294967295 sambaPwdCanChange: 1161193814 sambaPwdMustChange: 4294967295 sambaLMPassword: xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx...

Hi. I'm having a problem with hashing passwords for use in the menu. If I specify plain passwords in my config it works file: Example: MENU MASTER PASSWD 1234567890 MENU PASSWD test123 However if I hash a password using the sha1pass script the resulting hashes don't work. MENU MASTER PASSWD $4$9qj4qv8g$HQ6Jl6TVrpign78XeofX2OLmfJo$ MENU PASSWD test123 $4$B4LW6NPr$xYYfSoakhfLpWE7+l8zU0MdAlIA$ When prompted for the password I enter that which I...

...nks! (IP's and ISP and phone number disguised) - Executing [s at macro-dialexternal:57] GotoIf("IAX2/S-14468", "1?dialnormal") in new stack -- Goto (macro-dialexternal,s,60) -- Executing [s at macro-dialexternal:60] Dial("IAX2/S-14468", "IAX2/ISP123/1234567890|60|W") in new stack -- Called ISP123/1234567890 -- Call accepted by 201.191.37.138 (format ulaw) -- Format for call is ulaw -- IAX2/ISP123-2261 answered IAX2/S-14468 -- Channel 'IAX2/S-14468' ready to transfer -- Channel 'IAX2/ISP123-2261' ready to trans...

...;, -1 }, @@ -77,8 +89,14 @@ test_nbdkit_parse_size (void) { "1.5M", -1 }, /* Valid strings */ + { "-0", 0 }, { "0", 0 }, + { "+0", 0 }, { " 08", 8 }, + { "1", 1 }, + { "+1", 1 }, + { "1234567890", 1234567890 }, + { "+1234567890", 1234567890 }, { "9223372036854775807", INT64_MAX }, { "1s", 512 }, { "2S", 1024 }, @@ -88,12 +106,14 @@ test_nbdkit_parse_size (void) { "1K", 1024 }, { "1m", 1024 * 10...

...locks their number in outgoing calls. ====== Here's the command-line PHP script: #!/usr/bin/php <?php $fh = fopen('/root/output.txt', 'w'); fwrite($fh, "Received " . $argv[0]); fclose($fh); ?> ====== Here's extensions.conf: exten => 777,1,Set(CALLERIDNUM=1234567890) exten => 777,n,ExecIf($[${LEN(${CALLERIDNUM})} = 10],AGI(/root/dummy.php),${CALLERIDNUM}) ======... and here's the console :-/ -- Executing [777 at my-phones:1] Set("SIP/9001-088aa918", "CALLERIDNUM=1234567890") in new stack -- Executing [777 at my-phones:2] Ex...

:) How?? I know that I could convert unix time to standard time like: date -d "@1234567890" but how can I convert standard time to unix time? :D thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20090618/3eb12736/attachment-0001.html>

...ed to generate the data in the users sambaSID LDAP entry. As a result, The "Owner" field in the Windows file Explorer is correct for the homes share from the PDC in that it displays the "workgroup\joeuser" posix name. Shares mapped from the file servers show the long S-1-5-21-1234567890-1234567890-1234567890-2308 (with the last four being the rid?) where the first part is the SID from the PDC as the "Owner". Is there a way to make these display the right posix name also instead of the long non-useful sambaSID? Thanks!! Mike

...;, -1 }, @@ -77,8 +89,14 @@ test_nbdkit_parse_size (void) { "1.5M", -1 }, /* Valid strings */ + { "-0", 0 }, { "0", 0 }, + { "+0", 0 }, { " 08", 8 }, + { "1", 1 }, + { "+1", 1 }, + { "1234567890", 1234567890 }, + { "+1234567890", 1234567890 }, { "9223372036854775807", INT64_MAX }, { "1s", 512 }, { "2S", 1024 }, @@ -88,12 +106,14 @@ test_nbdkit_parse_size (void) { "1K", 1024 }, { "1m", 1024 * 10...

...team just because I believe it's an issue of the GROUP information being released when Asterisk consolidates the channels (removes all the MASQ channels) once the call is answered. But maybe it's something else so I'll ask here first. The dialplan setup: exten => 555,1,Dial(Local/1234567890) exten => _NXXNXXXXXX,1,Macro(which-line,${EXTEN}) [macro-which-line] exten => s,1,set(GROUP()=${DIALSTRING}) exten => s,n,Dial(${DIALSTRING}/1${ARG1}) Things are a bit more complex, but it's all just logic. The extensions above should give a decent representation of what's going...

...echo cancellation off but it makes no difference. This is what I get in /var/spool/hylafax/log: [root at faxhost log]# cat c000000003 Jan 14 12:44:43.39: [ 3403]: SESSION BEGIN 000000003 18005551212 Jan 14 12:44:43.39: [ 3403]: HylaFAX (tm) Version 4.4.6 Jan 14 12:44:43.39: [ 3403]: CallID: "1234567890" "/var/spool/asterisk/fax/20100114124443-1234567890-08451234567" "<NONE>" "08451234567" Jan 14 12:44:43.39: [ 3403]: MODEM set XON/XOFF/FLUSH: input ignored, output disabled Jan 14 12:44:43.39: [ 3403]: <-- [4:ATA\r] Jan 14 12:44:47.11: [ 3403]: --> [7...

...ame SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.example.com client_uri=sip:1234567890 at sip.example.com retry_interval=60 contact_user=siptrunk-in [siptrunk-in] type=endpoint transport=udp-transport context=from-trunk disallow=all allow=ulaw outbound_auth=siptrunk aors=siptrunk identify_by=uri Registration section has option "contact_user". Incoming call from this reg...