similar to: Remove "name" part of SIP From header

Hello I recently have got my setup up and running. I am using 2 x100p cards in a Dell 400SC. I get static (inbound and outbound) within the first 20 seconds on one card, but on the other card it is crystal clear all the time. They both have their own irq and I have tried to remedy the problem by doing a hdparm found in the wiki, but still nothing. I've tried rxgain and txgain, but that

Hi, I've been noticing a lot of these messages lately: "NOTICE[10235]: sched.c:220 ast_sched_add_variable: Scheduled event in 0 ms?" Is something broken? I'm running asterisk-1.4.22.1. They seem to happen in a number of different places where a beep or recording is played, such as when someone leaves voicemail or when an AGI script I have plays a time announcement -- lots

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 digits, i.e. the minumum, in "e" notation) to

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()

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:

I have got call recording working on our 1.4.30 asterisk box together with a recording pause ability and being able to play different audio to each party at the start and end of the pause. This all works perfectly but one wish is to have the audio files have a beep or something in them so when you listen later you can tell where the audio was paused. So I changed things around so that instead

Hi all, I was wondering if it is possible to print the callerid value in the CLI when doing 'pri debug span 1' For example > Call Ref: len= 2 (reference 2707/0xA93) (Terminator) > Message type: CONNECT (7) > [18 03 a9 83 97] > Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 > ChanSel: Reserved >

Hi: I am useing asterisk 11.12. I use G722 as preferred codec for my ip-phone. and my PSTN DAHDI use alaw. G722 is great when ip-phone talks to each other. but when ip-phone dialout to PSTN DAHDI, G722 is not great, since it is need to transcode to alaw. so I try to change the codec when dial from SIP to DAHDI. I tried to use IP_CODEC/SIP_CODEC_OUTBOUND at dialplan. but the SIP

Gentlemen, Did tryout "CONNECTEDLINE" function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and "stupid" extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from "955" to "Connected Line 955" when my call is answered, shouldn't the

Hello, I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled. When I receive a SIP INFO, the logs tell me that a "DTMF begin" is generated, but no related "DTMF end" is generated, unless the call is ended. Here is an excerpt of the logs : *--- SIP INFO received **on **SIP/xxx-00000004:* [Dec 13 11:56:16] DTMF[18193][C-00000005]

Hi, After lots of struggle and rtfm I finally got most things running, except for 'nmblookup' and 'net getlocalsid' on the BDC. I'm not new to Samba, but plenty more to learn. Here's the setup in summary: system pdc is the PDC on subnet 192.168.0.0, running SuSE10.1, LDAP master, wins server, domain master browser, no iptables; system bdc is the BDC on subnet 192.168.2.0,

https://bugs.freedesktop.org/show_bug.cgi?id=107963 Bug ID: 107963 Summary: kernel rejected pushbuf: Invalid argument Product: xorg Version: unspecified Hardware: Other OS: All Status: NEW Severity: normal Priority: medium Component: Driver/nouveau Assignee: nouveau at

Package: xen-hypervisor-4.0-i386 Version: 4.0.1-4 This is the same machine referenced in http://bugs.debian.org/665413. It is an HP d530 SFF workstation (model DG784A) with 4GiB of RAM. It has run for years on the lenny xen + linux stack. If my notes are correct, it also ran smoothly for about a year with the lenny hypervisor + kernel and the squeeze userland. Rebooting into squeeze xen +

Hi, I've migrated from an old samba installation (Samba as PDC) that used TDB backend for password. I've setup a box with ubuntu and samba 3 + ldap and I imported the old users. Old users works fine. I have problems with new users and machines. Old users works but they don't show up with smbldap-usershow command and I've problem in changing their passwords. If I check the ldap

Hi, I am getting a crash in nouveau in my application. It's basically a java application, and I am loading a bitmap into an opengl texture and showing it in a panel. Below is a snippet of the console output. Sometimes I will get a lock up instead of a crash. My whole desktop will lock up, but I can still move the mouse pointer around. Please let me know if you need more information. Your

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

I am trying to connect an IAX ATA to an Asterisk 1.4.21.2 system. The Asterisk system has been stable for years, and has no trouble bridge SIP phone sets to IAX trunks. When I initiate a call from the IAX ATA, something goes wrong. One rare occasion it works fine, but usually there is no audio passed. I have a snippet of the console below. Notice no bridging message...not sure if that's

https://bugs.freedesktop.org/show_bug.cgi?id=107829 Bug ID: 107829 Summary: nouveau crash/freeze [MULTIPLE_WARP_ERRORS] warp 3f0009 [ILLEGAL_INSTR_ENCODING] Product: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: major Priority:

Dear, i using this scenario. jitsi---> asterisk---->EPABX------> Local Telephone when i am calling from jitsi to no 88 its giving this message and getting busy tone. == Using SIP RTP CoS mark 5 -- Executing [88 at myphones:1] Dial("SIP/sandeep-00000004", "DAHDI/g0/88,20,rt") in new stack -- Called g0/88 [Nov 15 09:53:54] WARNING[3169]: chan_dahdi.c:7536

Hi All, I'm using the asterisk 1.4.39.2 with phpagi 2.20 I have setup a dial plan: [callback-outbound] exten => _00.,1,Macro(callout|${EXTEN}) [macro-callout] exten => s,1,AGI(getchannel.php|${ARG1}) exten => s,2,Dial(Local/${OUTBOUND}@from-internal/nj||tr) exten => s,3,Hangup() but for some reason i am not receiving the argument: Executing [s at macro-callout:2]