similar to: Bypass local dialplan and redirect INVITE

Is there a way to let chan_sip skip host lookup? Problem is I have to have a peer host config for every sip message outgoing. For example, I cann't have this in extension.conf exten => 500,n,Dial(SIP/romi at 192.168.1.79) It'll return, chan_sip.c:2738 create_addr: No such host: 192.168.1.79 when call forwarding I have to have a peer in SIP [outgoing] host=192.168.1.79 ... in

When making an outbound call, the outbound peer return a 301 forwarded with URI to other domain, but asterisk think it's a local domain and try to look it up from extension.conf. How to configure so that a 301 forwarded with URI from other domain thinks it's outgoing to another proxy? thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello. I would like to redirect some ports from user network to host system. In console i can do it: /usr/libexec/qemu-kvm -drive file=/home/libvirt/qemu-fedora.img -vnc 10.200.1.1:1 -net user -redir tcp:22000::49999 -redir tcp:22001::22 How can i do it with libvirt? I make users network in XML config VM: <interface type='user'> <mac

Hi all, See the following commands: werner at debian:~$ rsync -c ftp.cn.debian.org::debian/ 2 >aaa rsync: The server is configured to refuse --checksum (-c) rsync error: requested action not supported (code 4) at clientserver.c (849) [sender=3.0.9] rsync: read error: Connection reset by peer (104) rsync error: error in socket IO (code 10) at io.c(785) [Receiver=3.1.2dev] Why rsync cann't

Hongyi Zhao <hongyi.zhao at gmail.com> (Mi 15 Apr 2015 13:32:23 CEST): > Hi all, > > See the following commands: > > werner at debian:~$ rsync -c ftp.cn.debian.org::debian/ 2 >aaa > rsync: The server is configured to refuse --checksum (-c) > rsync error: requested action not supported (code 4) at clientserver.c > (849) [sender=3.0.9] > rsync: read error:

I want to setup a blind transer for auto forwarding to SIP peer. I have context forwarding looks like this in extension.conf [forwarding] ... exten => 511,1,Dial(SIP/sip_proxy-out) ... This will do the re-invite, which is attendance transfer maybe. But I want a blind transfer by REFER method. How can I do that? I know that the transfer() function may be able to do that. But I don't know

Hi all, My team and I are finding ourselves a little in the dark about the "CSRF Protection Bypass in Ruby on Rails" vulnerability that was announced yesterday - http://weblog.rubyonrails.org/2011/2/8/csrf-protection-bypass-in-ruby-on-rails 1. Where is the complete Advisory? The Impact section is very unclear. Looking at the comment in the 2.3 patch mentions "Flash animations and

https://bugzilla.samba.org/show_bug.cgi?id=1599 Summary: copy-unsafe-links cann't take affect Product: rsync Version: 2.6.2 Platform: x86 OS/Version: Linux Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned@samba.org ReportedBy: faris.xiao@haoxi.com

rhbz#:51422 --- ovirt-node.spec.in | 149 +++++++++------------------------------------------ 1 files changed, 27 insertions(+), 122 deletions(-) diff --git a/ovirt-node.spec.in b/ovirt-node.spec.in index 3138011..b4e660d 100644 --- a/ovirt-node.spec.in +++ b/ovirt-node.spec.in @@ -43,76 +43,23 @@ Requires: nc Requires: grub Requires: /usr/sbin/crond Requires: anyterm

Hi All, I use the Asterisk Manager Interface to redirect the channels. There have two channels : SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456) SIP/612-5456 s@macro-monitor:10 Up Dial(SIP/0882@voip_out Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel:

Hello list, is it possible to extract the Remote-Party-ID from an incoming call in the dialplan ? Is there some kind of function for this ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110316/bda02b4d/attachment.htm>

Hi, I have a requirement where an IVR application on asterisk has to play a audio file in g729 and when a digit is pressed, the call should switch to another codec (say ulaw). So, What can I do in the extensions.conf to trigger a re-negotiation of codec? I used exten => 55xx,n,Set(SIP_CODEC=ulaw) but, I suppose this affects the next call and not the current one. Please help ASAP Thanks,

I have Samba PDC and use tdbsam backend. I use the version 3.0.11, but recently I have exchanged it for the version 3.0.23c. Now I have a number of problem. Here is one of then. I wish to move one user "gad" from Domain Users (RID 513) to Domain Guests (RID 514). Now Primary group of user "gad" is "Domain Users": pdbedit -Lv gad -d0 Unix username: gad NT

On Sun, 17 Dec 2017 21:48:05 +0800 weiping zhang <zwp10758 at gmail.com> wrote: > As mentioned at drivers/base/core.c: > /* > * NOTE: _Never_ directly free @dev after calling this function, even > * if it returned an error! Always use put_device() to give up the > * reference initialized in this function instead. > */ > virtio_register_device may fail before/after call

On Sun, 17 Dec 2017 21:48:05 +0800 weiping zhang <zwp10758 at gmail.com> wrote: > As mentioned at drivers/base/core.c: > /* > * NOTE: _Never_ directly free @dev after calling this function, even > * if it returned an error! Always use put_device() to give up the > * reference initialized in this function instead. > */ > virtio_register_device may fail before/after call

Hi: I encounter a promble when I use libvirt manager my virtual machine. I use dhcp allocate IP for my guest, but use dhcp manager IP source, I will ensure guest run NetworkManager auto start NIC in dhcp type get IP. But some release system no install NetworkManager, or some installed system no install NetworkManager, so when I manager those guest, I cann't allocate IP for guest via dhcp

Hi, After a whole day of work, I finally complied oh323 0.6.3a successfully. But when I started asterisk, it cann't load oh323. Following is the error: [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [cdr_csv.so] => (Comma Separated Values CDR Backend) [chan_oh323.so]Jul 13 09:43:45

Hello I'm learning more about dialplans and have a couple of questions: 1. Am I right in understanding that the actions that can be performed in extensions.conf can be of two types only: - internal commands (Dial, Wait, etc.) - calls to external scripts throught AGI? 2. I'd rather write scripts in Python instead of Perl or PHP. Does someone have a skeleton that I could use to build one,

Hi everyone: I use dovecot only as the imap server, For smtp and pop3, I use apache james. The dovecot version: 1.1.14 And the configuration is : protocols: imap ssl_disable: yes login_dir: /var/run/dovecot/login login_executable: /usr/libexec/dovecot/imap-login mail_location: mbox:/usr/local/edupass/mail/repo/%u:INBOX=/usr/local/edupass/mail/%u auth default: passdb:

Hi all, I know when doing a Dial, when the called party hangs up, we have a few different ways to redirect the calling party to other parts of the dialplan. In this case, I have someone who would like to do the opposite... When the calling party hangs up after a Dial(), redirect the called party to another location. I'm not sure how else to describe what the user wants to do, but I'm