search for: 5678

Displaying 20 results from an estimated 178 matches for "5678".

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2010 Jul 28
2
Nat issue one way audio on IP dial
...tings in conf files to correct this problem. Below is the conf of calling user [pepsi] username=pepsi type=friend secret=123456 qualify=yes nat=no insecure=port,invite incominglimit=1 outgoinglimit=1 host=dynamic dtmfmode=rfc2833 context=out canreinvite=yes callerid="pepsi coke" <12345678901> accountcode=6:0:pepsi amaflags=default disallow=all allow=alaw allow=ulaw allow=g729 allow=gsm Below is the conf of called user [adf] username=adf type=friend secret=123456 qualify=yes nat=yes insecure=port,invite incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=user c...
2015 Jun 30
2
Call for testing: OpenSSH 6.9
...regression test because sshd -T # will fail if we're not running with SUDO (no permissions for real keys) or # if we are # running tests on a system that has never had sshd installed @@ -26,6 +31,9 @@ cat > $OBJ/sshd_config.0 <<EOD listenaddress 1.2.3.4:1234 listenaddress 1.2.3.4:5678 +EOD + +[ X${SKIP_IPV6} = Xyes ] || cat > $OBJ/sshd_config.0 <<EOD listenaddress [::1]:1234 listenaddress [::1]:5678 EOD @@ -37,6 +45,9 @@ port 1234 port 5678 listenaddress 1.2.3.4 +EOD + +[ X${SKIP_IPV6} = Xyes ] || cat > $OBJ/sshd_config.1 <<EOD listenaddress ::1 EOD...
2016 Feb 15
2
Asterisk 13.6.0/The simplest TCP configuration does not work
...3-1---d8754z-;rport;transport=TCP Transport: TCP Sent-by Address: 192.168.1.15 Sent-by port: 47053 Branch: z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z- RPort: rport transport=TCP Max-Forwards: 70 Contact: <sip:5678 at 192.168.1.15:47053 ;rinstance=bea6f11f37c55605;transport=TCP> Contact URI: sip:5678 at 192.168.1.15:47053 ;rinstance=bea6f11f37c55605;transport=TCP Contact URI User Part: 5678 Contact URI Host Part: 192.168.1.15 Contact URI Host Port...
2011 Aug 25
2
string manipulation
I R-users, I am trying to find the way to manipulate a character string to select a 4 digit number after some specific word/s. Example: mytext <- "I do not want the first number 1234, but the second number 5678" Is there any function that allows you to select a certain number of digits (in this case 5678) after a particular word/s (e.g., second number) Thank you for your help Lorenzo [[alternative HTML version deleted]]
2016 Feb 15
2
Asterisk 13.6.0/The simplest TCP configuration does not work
Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >> [Feb 15
2006 Nov 07
5
mechanize: 400 Bad Request
...But I don''t know, if this is the reason for the malfuncion and how to fix this. mechanize: Net::HTTP::Get: https://www.frankfurter-fondsbank.de/../diverse/navigation.jsp;jsessionid=CdEfG!-34567!-7654?menu=1 firefox-LiveHTTPHeaders: GET /diverse/navigation.jsp;jsessionid=AbCdE!-1234!-5678?menu=1 HTTP/1.1 Thank you for any help! Axel axel ? friedrich ? _smail AT gmx ? de Details ??????? ruby 1.8.4 (2005-12-24) [i386-mswin32] Windows 98SE Code ???? require ''rubygems'' require ''mechanize'' agent = WWW::Mechanize.new page = agent.get("h...
2010 Aug 03
2
RTP stream not passing through router with port forwarding
...server. asterisk server is on public ip so no port forwarding and natting necessary there. however caller and callee both are behind router and there is port forwarding enabled and nat=yes, qualify=yes in sip.conf for both users. callee user name: adf callee local ip/port: 192.168.0.10:5678 callee router ip: 116.79.x.x when we simply dial callee as Dial(SIP/adf) RTP stream reaches perfectly fine to 192.168.0.10 through router and INVITE is sent to local ip through router. INVITE sip:adf at 192.168.0.10:5678 SIP/2.0 (asterisk somehow manages to contact local ip through rout...
2005 Jan 31
0
Caller ID Bug in v1.0.5
...en I retransmit the request with the proxy auth, the "From" number becomes the "To" number in the SIP message. Here is an example (with ambiguous numbers): Reliably Transmitting: INVITE sip:1234@10.0.0.2 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK38d10979 From: "5678" <sip:5678@10.0.0.1>;tag=as18c39e1b To: <sip:1234@10.0.0.2> Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.0.0.1:5060;branch=z9hG4bK38d10979 From: "5678" <sip:5678@10.0.0.1>;tag=as18c39e1b To: <sip:1234@10.0.0.1>;tag=as35adb52e Prox...
2010 Jul 16
0
asterisk-users Digest, Vol 72, Issue 39
yes, actually this scenario is on remote servers. like SIP/XYZ at 119.18.230.20:5060 SIP/XYZ at 202.68.0.90:5678 audio is ok when dialing without using ip & port as below SIP/XYZ but when i dial using below dialstring SIP/XYZ at 202.68.0.90:5678 or SIP/XYZ at 119.18.230.20:5060 then the problem arises hope you got the idea.. Nasir ------------------------...
2020 Jul 10
2
RFC: Bugzilla migration plan
...> and starts from zero again? I will need to clarify whether we will be able to reset the counter or not > > 4. Migrate all issues from llvm-bugzilla-import to llvm-project using > > GH API. Github will take about llvm-bugzilla-import/issues/1234 => > > llvm-project/issues/5678 redirects > If we're setting a redirect, PR1234 wouldn't hit #5678. We either > guarantee that the IDs will be identical or we'll need a smart > redirect that will know the delta (or 1:1 relationship). Why? If you migrate the issue inside GH, then GH does the necessary redirect...
2015 Mar 12
2
chanspy for group extension
thank you so much it work you must add 1 like below [app-chanspy] exten => _0071XX,*1,*Macro(chanspy,1234) exten => _0072XX,*1,*Macro(chanspy,5678) exten => _0073XX,*1,*Macro(chanspy,8910) best regards. 2015-03-11 19:48 GMT+00:00 Carlos Chavez <cursor at telecomabmex.com>: > On 3/11/15 12:48 PM, Salaheddine Elharit wrote: > >> hello list, >> >> i use chanspy with the code below >> >> [app-chans...
2015 Mar 11
2
chanspy for group extension
...py(SIP/${EXTEN:3},dqs) exten => _007.,n,Hangup i have a question related to chanspy i have created extension from 100 to 300 and i will give the permission with group of extension i want to use chanspy like below 100=====>199 with Authenticate(1234) 200=====>299 with Authenticate(5678) 300=====>399 with Authenticate(8910) any help please Thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150311/19deb336/attachment.html>
2002 Jan 18
1
scp between two remote hosts
Hi All, I am running openssh (OpenSSH_2.9p1, SSH protocols 1.5/2.0). The scenario is that I have got three machines (A, B and C). The sshd on host A is listensing on port 1234, and the sshd on host B is listensing on port 5678. How can I set up a scp from a third host C so as to copy a file from host A to host B? scp -P 1234 myname at A:/var/tmp/file1 -P 5678 myname at B:/var/tmp/file1 does not work and C always try to connect to sshd on A at port 22. Rgds. Bob Shih PLEASE READ: The information contained in this em...
2019 Apr 17
2
IPv6 transport results in ICE with only IPv6 candidates
Hi, I'm using Asterisk 13.x and have defined a pjsip TCP IPv6 transport: [transport-tcp-ipv6] type=transport protocol=tcp bind=[2001:1234:5678:abcd::2]:5060 I also have an IPv4 version of that: [transport-tcp-ipv4] type=transport protocol=tcp bind=10.75.22.8:5060 I've then configured an endpoint to use it: [outgoing] type = endpoint context = default dtmf_mode = none disallow = all allow = all rtp_symmetric = yes force_rport = yes...
2011 Aug 29
2
Dialing multiple endpoints and CallerID presentation
...ough the DAHDI span, must have CallerIDs presented without any prefix. Ideally, CallerID should be presented : 1- with 4-digits for internal phones 2- with 10-digits for external phones so that both phones can return the call without re-dialing. Suggestions ? A is 1234 alias DID 0555551234 B is 5678 C is 0123456789 I was thinking of using something like this: Dial(SIP/5678<option_to_present_1234_to_callee>&DAHDI/g1<option_to_present_0555551234>/0123456789) What could be <option_to_present_1234_to_callee> and <option_to_present_0555551234> Regards -------------- n...
2020 Jul 10
7
RFC: Bugzilla migration plan
...orage format. 2. Install redirects llvm.org/PR1234 => gh/llvm/llvm-bugzilla-import/issues/1234 3. Wipe existing issues and pull requests 4. Migrate all issues from llvm-bugzilla-import to llvm-project using GH API. Github will take about llvm-bugzilla-import/issues/1234 => llvm-project/issues/5678 redirects The only downside of this approach is that we will be seeing 30k events like "llvm-bugzilla-import/issues/1234 migrated to llvm-project/issues/5678". Here is the tentative timeline / list of action points: 1. Collect the mapping email (used by bugzilla) => GH account name...
2015 Mar 12
0
chanspy for group extension
hello list, i use the code below [macro-chanspy] exten => s,1,Authenticate(${ARG1}) exten => s,n,ChanSpy(SIP/${EXTEN:3},dqs) exten => s,n,Hangup app-chanspy] exten => _0071XX,*1,*Macro(chanspy,1234) exten => _0072XX,*1,*Macro(chanspy,5678) exten => _0073XX,*1,*Macro(chanspy,8910) but when i do 007100 for exemple i spy another agnet 102 or 103 any help please thanks and regards 2015-03-12 10:30 GMT+00:00 Salaheddine Elharit <salah.elharit200 at gmail.com>: > thank you so much it work > you must add 1 like below...
2006 Jul 21
1
Unable to configure squid transparent proxy on Centos4.0
...26/24 Debian: eth0: 192.168.2.83/24 gateway: 192.168.2.126/24 (eth1 of Centos) Squid works fine if I manually add the proxy settings in any browser for the clients of network (192.168.1.0) but as I want to configure transparent proxy I have added these lines to squid.conf http_port 5678 httpd_accel_host virtual httpd_accel_port 80 httpd_accel_with_proxy on httpd_accel_uses_host_header on iptables configuration Only one rule is there iptables -t nat -A PREROUTING -p tcp --dport 80 -j REDIRECT --to-port 5678 I can access the LAN servers fr...
2005 Jan 06
1
Problems with MeetMe accepting conference PIN
...cepting a PIN number for a conference room. At this point in time I have established the conference definition in the meetme.conf file as well as specifying the appropriate lines in the extensions.conf file. meetme.conf file: conf => 1234,1574 extensions.conf file: exten => 1234,1,MeetMe(5678) OR exten => 1234,1,MeetMe,5678 In both cases I can get the SIP client "SJPhone" to actually dial the conference number and I hear the operator say, enter the pin number followed by a "pound sign". After entering in the PIN number and entering in the HASH button, the sys...
2015 May 29
0
Calling from "extern"
Hi list! Finally I got my wife's phone working in my Asterisk. Unfortunately I have some problems, too... Current situation: - AsteriskNOW with 4 Accounts (00493511111111, 00493512222222, 00493513333333, 5678). This is "for test" and it will be replaced by "the real world", when I got my Asterisk to work... - A second Asterisk (Ubuntu-PBX) on another VM, logging in AsterikNOW **AND** Messagenet - 2 VoIP phones, logged into Ubuntu-PBX (my phone, my wife's phone) - A Twinkle inst...