similar to: Uprading to Asterisk 11 issues

Originally filed by Robert LeBlanc as Debian Bug # 652679 - <http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=652679> <Quote> Package: winbind Version: 2:3.6.1-3 Severity: important Dear Maintainer, After upgrading to 3.6.1 I am no longer able to login to Debian using my Active Directory account. 'winbind -u', 'winbind -g', 'winbind -t' and many others work

I have both the PJSIP add and the chan_sip way of adding SIP headers in there. The Verbose is showing the variable value is there. The INVITE to PJSIP/Agent1 does not include either X-My-DNID or X-My-DNID2 headers. exten => 1234,1,Verbose(X-My-DNID:${MY_DNID}) same => n,Set(X-My-DNID=${MY_DNID}) same => n,Set(PJSIP_HEADER(add,X-My-DNID2)=${MY_DNID}) same => n,Dial(PJSIP/Agent1)

Ishfaq Malik <ish at pack-net.co.uk> schrieb: Hi Ishfaq > Look into Busy Lamp Field/Presence > > Here's a starting point: > > http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DeviceStates-SECT-1.html Thanks a lot, but it does not seems to work... Here my configuration: sip.conf: [general] allowsubscribe=yes subscribecontext = default

READ READ READ .... http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-DP-Basics.html Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Sun, Feb 22,

Thanks Scott. I was able to get the basic concept to run. However, it seems PJSIP INVITE for the Dial also does not support added headers. The Local channel dial plan did have the channel variable values. I added them as SIP headers, then Dial(PJSIP/Agent). The INVITE for the Dial on PJSIP continues to not include the SIP Headers I added. For chan_sip, I have no problem with this. Even the

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

I am trying to find where the authoritative sql definitions for Asterisk Realtime Architecture ARA are located. I have found many locations but each and everyone seems to be different. http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example Files included with the distribution:

Hello all, I am going to register asterisk sip users through active directory accounts LDAP (that is a separated server with ip : 192.168.11.17) So I have followed the below link as well: https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html

(I'm new to Asterisk, after having started VOIP with vat on the mbone in the 90s.) I am setting up my first Asterisk system, and trying to read docs/guidance and follow best practices. I have read the 5th Edition of "Asterisk: The Definitive Guide" and like the 3rd Edition on the web it recommends that hardphones and softphones both have a unique name distinct from any concept of

Hello, I have been using successfully Dovecot 1.1.x for about a year now. It has been very stable. Now I'm uprading that same system to newer and more powerful hardware and I was wondering whether it is good idea or not to switch to Dovecot 1.2.x series. Could anybody direct me to feature comparision document or explain here main differences betweeen thos two branches? -- Veiko

I am wondering if any one knows of a way to manage Xyratex disk shelves from CentOS (in particular CentOS 4). More details: Some years ago I installed a NAS unit from Exanet which consists of 2 rebadged IBM x3650 head nodes and a couple of Xyratex disk shelves with a total of 96 TB of raw disk, connected by fibre channel. The operating system is based on CentOS 4.4, but is modified, and runs a

Thanks a lot for the reply. I did follow that already, but i do have a problem. Here is my extensions.conf part for that particular number exten => 6912345678,1,Answer() exten => 6912345678,n,MYSQL(Connect connid 127.0.0.1 root mypasswd asterisk) exten => 6912345678,n,MYSQL(Query resultid ${connid} SET NAMES utf8) exten => 6912345678,n,GotoIf($["${connid}" =

Hi, I am trying to create a VM with the IVSHMEM feature, and I specify the device using qemu command line argument in my XML by using the pass through tag as follows: <qemu:commandline> <qemu:arg value='-device ivshmem,size=2048M,shm=fd:/mnt/huge/map_1:0x0:0x40000000:/dev/zero:0x0:0x3fffc000:/var/run/.ivshmem_metadata_config_2:0x0:0x4000'/>

I am using asterisk as SIP/GSM gateway. I have 2 gsm cards installed in server. I am having some issue in audio quality. I want to enable jitter buffer on asterisk but don't know, how to do. Any one can help me. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130423/015bd0de/attachment.htm>

Hello; How I can compare between Asterisk 1.8 and 11 with reference to the following points: 1) SMS. 2) gtalk and other social media. 3) GUI. 4) Any main difference? Regards Bilal

Hi, is possible that two sip extensions: user-1 and user-2 are connected and I want to transfer the call from user-1 to a third user "user-3". I know it is possible through feature keys mapping in features.conf, but I want to do this through AMI or Asterisk CLI Commands? Please suggest if possible? Thank you! Muhammad Faheem -------------- next part -------------- An HTML attachment

Hi, i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing "XXX" -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I am using javascript to call an action with some parameters. In that action i decide on the basis of parameters either to render some text on that page or to re direct to some other page. But its seems that my redirect_to method is not working from that action. Can anybody tell me some way of redirecting to some other action when that action is called as a remote_function from

Hi all, We're playing around with new versions and uprading options. We currently have a 2x2x2 stripped-distributed-replicated volume based on 3.3.0 and we're planning to upgrade to 3.4 version. We've tried upgrading fist the clients and we've tried with 3.4.0, 3.4.1 and 3.4.2qa2 but all of them caused the same error: Failed to get stripe-size So it seems as if 3.4 clients are