search for: 1234567

I am currently attempting to split a long list of strings (let's call it "string.list") that is of the format: "1234567.z3.abcdef-gh.12" I have gotten it to: "1234567" "z3" "abcdef-gh" "12" by use of the strsplit function. This leaves me with each element of "string.list" having a split string of the above format. What I'd like to do now is extract the...

...again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk replied to all the INVITE's it received before it says Ringing. Really need help on this badly, anyone has an idea. Thank you in advance. Regards Ron U 172.16.0.6:5068 -> 172.16.0.1:5060 INVITE sip:1234567 at 172.16.0.1 SIP/2.0..Via: SIP/2.0/UDP 172.16.0.6:5068;branch=z9hG4bK1531e34025a31be2..From: ""<sip:unknown at 172.16.0.1>;tag=57677d009236 ed33..To: <sip:1234567 at 172.16.0.1>..Contact: <sip:172.16.0.6:5068>..Supported: replaces, timer, path..Call-ID: 02075d60f...

Dear R experts, format(12345678, digits = 2) gives [1] "1.2e+07" while format(1234567, digits = 2) gives [1] "1234567" but I'd like the last number to be represented as "1.2e+06" string too. Where am I wrong? Thanks, Timur.

Hi I have a 3 level nested while-endwhile loop in a macro that when the execution reaches endwhile, it is jumping out to the While at the caller macro. It shouldn't since the are instructions after the endwhile. -- Executing [s at macro-call-from-outside:72] EndWhile("DAHDI/i1/1234567-4a7f", "") in new stack == Channel 'DAHDI/i1/1234567-4a7f' jumping out of macro 'call-from-outside' -- Executing [s at macro-recurse_check_redirect_not_mailbox:7] While("DAHDI/i1/1234567-4a7f", "1") in new stack I checked the while-endwh...

Hi folks! Is it possible to tell asterisk not to strip the leading 0 of *incoming* MSNs? I use asterisk with i4l and whenever I get a call from an long-distance party, the leading 0, which should be there according the german numbering, is not. So if I get a call from a mobile phone 0177-1234567 should be displayed, but 177-1234567 is displayed. I double checked if I've forgotten to remove an option to strip the first digit of incoming calls and found nothing. The wiki and the mailinglist archives can't enlight me either, why asterisk behaves like this, or how I can turn it off. S...

...with incoming calls from sipgate. This was working in 1.4 but in 1.6 I get a 401 Unauthorized :-(. Sipgate has mentioned that I have to change the type to friend, but it is already friend, so what's wrong? Kind regards, Michael Here is the sip.conf: [sipgate_out] type=friend nat=yes username=1234567 fromuser=1234567 fromdomain=sipgate.de secret=secret host=sipgate.de qualify=yes Here is the SIP trace: <-------------> --- (18 headers 19 lines) --- == Using SIP RTP CoS mark 5 Sending to 217.10.79.9 : 5060 (no NAT) Using INVITE request as basis request - 40fa80331421fa800e4633bd497e1cd8...

I have asterisk with txfax rxfax modules. In default contex I have: [default] exten => 1234567,1,rxfax(/home/patryk/fax-new.tif|debug) Rest o of configuration is default. I'm injecting this: Channel: Local/1234567 MaxRetries: 1 WaitTime: 4 Application: txfax Data: /home/patryk/fax.tif|debug|caller to var/spool/asterisk/outgoing/fax.call but I can't see incoming i fax in my home...

...Congestion() exten => _91877NXXXXXX,102,Congestion() exten => _91866NXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _91866NXXXXXX,2,Congestion() exten => _91866NXXXXXX,102,Congestion() exten => _91900NXXXXXX,1,Congestion() exten => _91976NXXXXXX,1,Congestion() exten => _91[1234567]XXNXXXXXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1}) exten => _91[1234567]XXNXXXXXX,2,Congestion() exten => _91[1234567]XXNXXXXXX,102,Congestion() exten => 9911,1,Dial(${OUTBOUNDTRUNK}/911) exten => 9411,1,Dial(${OUTBOUNDTRUNK}/411) exten => 0,1,Dial(${OUTBOUNDTRUNK}/0) [local-access] in...

Hi Everyone. Some time ago, I was told that it's possible to implement an incoming Fax server with extensions using only one PSTN line, like this: PSTN number: 1234567, that's the line connected, FAX numbers like: 1234567-00, 1234567-01, 1234567-02 and so on are routed to their respective recipients. I do not want real DID (which would mean buying a bunch of numbers from the telco, expensive), but to handle this inside my Asterisk PBX. I was told there is s...

I'm new to Asterisk, but, Managed to get it working for outound calls from my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In fact, voice mail won't even work.. This is a snippet of what I'm getting when I try to call the ATA -- Executing

...ing to send a tif file, using Fax for Asterisk and the call is executed, but when I get the reINVITE with T.38 data, the local server doesn't recognize that we have this capability and sends a 488 message. These are the logs: <--- SIP read from UDP:xxx.xxx.xxx.xx8:5060 ---> INVITE sip:1234567 at 10.0.0.3:5060 SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xx8:5060;branch=z9hG4bK0dB004c7e5e3c3e60a8 From: <sip:98765432 at xxx.xxx.xxx.xx8:5060>;tag=gK0d817deb To: "Fax" <sip:1234567 at yyy.yyy.yyy.yyy>;tag=as0ddeacb5 Call-ID: 74ca1e4e3e86a1b873428773477e201f at yyy.yyy.yyy.yyy C...

Hi can someone help me, I want to do 'Dial(IAX2/bla/1234567|50|tT)'

...and testing it with soft phone like sj-phone and x-lite without any serious problems, but: + I don't understand some settings. - what's deflect in capi.conf for? - how to configure multiple isdn-cards in capi.conf? + Some probs with the extensions.conf - exten => _0.,1,Dial,CAPI/@1234567:${EXTEN}|30|r is leading into Session rejected: 403 Forbidden exten => _0.,1,Dial,CAPI/@1234567:${EXTEN} works - we have a connection internally to another PBX via ISDN _*.,1,Dial,CAPI/@1234567:${EXTEN} or _9.,1,Dial,CAPI/@1234567:${EXTEN} to one special number to a call ce...

I'm using Asterisk a SIP Server for a lot of GrandStream HandyTone ATA's. Each one of them is configured in sip.conf as: [1234567] type=friend username=1234567 secret=1234567 callerid="ATA 1234567" host=dynamic nat=yes qualify=yes disallow=all allow=g729 canreinvite is set globally to YES. When one ATA calls another, I see the next traffic on Ethereal (just shown the sequence between one ATA and Asterisk) 1.- AT...

Hey all, I am trying to get a modified hostname (converting "blah-1234567" to "1234567") to use in my manifests and templates. I have it working for templates by using Ruby (*<%= hostname.split(''-'').pop.downcase %>*) but I haven''t been able to use it in manifests because they don''t process ERB. The way I see it I...

Hi... I noticed a bug with samba 2.2.8a on my SuSE 9.0 box. I use the UltraEdit text editor to edit my php files in a samba share. By default this text editor places .bak files next to the changed file. What I now noticed was this problem: I edit a file 1234567.php and it generates a file 1234567.php.bak. I can open and edit the 1234567.php normally. I edit a file 123456789.php with the same result as the 1234567.php. But: Once I edit a file 12345678.php and the 12345678.php.bak file is generated, the text editor no longer can access the 12345678.php....

...zip file is then deleted. I need to remove the old versions of the unzipped directory using Puppet when we rol out a newer version. So, on my client/node machine I have the below directories. /opt/1234 /opt/12345 /opt/123456 What I need to do, is rm the the above directories when we roll out 1234567.zip. When the newest directory (1234567) is unzipped, the Puppet exec sym links it to /opt/java. So could I remove the older directores that are not sym linked? Maybe using: tidy { tidy_opt: path => "/opt/", age => "1d", backup => f...

Hello, I'm running version R x64 v2.12.2 on a 64bit windows 7 PC. I'm trying to read a text file using read.table where the values have a format like "1,234,567". What I want is "1234567". Is there a quick way to strip out the commas? I can use strsplit and paste, but the file is quite large and would take some time. Thanks. Eric [[alternative HTML version deleted]]

hi all! I'm running an Asterisk-box with bristuff-....RC8n and 2 HFC-S cards. I m located in Vienna/Austria. I have the problem that on outgoing calls i hear my voice as a short echo (about half a second). This occurs not on every call. I tried some changes in my zapata.conf, with rxgain and txgain settings, but to me its hard to find a configuration which is good for every call i make. Is

...ame thing with both VoicePulse and another * server I have. [voicepulse] type=peer host=gw5.voicepulse.com trunk=yes user=USERNAME pass=PASSWORD and in my dialplan: Dial(IAX2/voicepulse/${EXTEN:2}@VPWS,90,r) The log shows this: -- Executing Dial("Zap/2-1", "IAX2/voicepulse/14121234567@VPWS|90|r") in new stack -- Called voicepulse/14121234567@VPWS WARNING[81926]: File chan_iax2.c, Line 4528 (socket_read): I don't know how to authenticate wSS36eLJ68 to 66.234.228.132 -- Hungup 'IAX2[voicepulse]/16384' wSS36eLJ68 is *not* my voicepulse username. Same th...