similar to: IP to IP call without server?

Hello! Does anybody know of a way to access the Request URI in a SIP message? I've got the following problem/scenario: We have a SIP Proxy (SER) wich forwards SIP-messages for non-IP destinations to our Asterisk. There is no authentication done between Asterisk and SER. I've configured Asterisk to accept any request for a PSTN-line from SER's IP-address. Since we allow IP-to-IP

Hello, I cannot accept any inboud calls from any provders in my asterisk which tries to authenticate the provider and at the end rejects the call with tthese message 407 Proxy Authentication Required How do I turn off this message. Thanks. Ehsanul Karim

Hi, I'm running asterisk-1.0.2-2mdk. When I tried to start it with /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvgc, I get [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) Ouch ... error while writing audio data: : Broken pipe # ps aux | grep mpg123 root 5237 0.1 0.4 5816 4444 pts/0 S 18:45 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3

Vacation (Sieve) replies to the envelope address. However, I have a customer which receive e-mails from a service which sends e-mails on behalf of a user (submitted through a web form). So an e-mail would be like (envelope from and from-header is webformfromwhatever at foobar.com): From: webformfromwhatever at foobar.com To: mycustomer at hisdomain.com Reply-To: usersrealmailaddress at

I am trying to "dial" a GTalk, ie @gmail.com, address. I inscribed this address in jabber.conf on the buddy= line. Upon executing the Dial application, I hear only a brief brief ring, then nothing. What might be the trouble? As the Dial application starts trying, the JABBER chatter on the console includes some "INCOMING" entries that name IP addresses. The one with

Hey all, Does anyone know what this means? I was running asterisk fine. Installed it on a new pc and I am using the g729b. codec that is optional. I ran the install for the codec it went ok but when I run askterisk via asterisk -vvvgc it gives me this error anyone know? I make sure I entered in the correct reg number. I followed the steps correctly. Too Registration error! Please try

Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, It can't monitor any lines and I have to restart the phone to reactivate this function. Is this a specific problem of asterisk-1.2.3? How can I solve it? Thank in advance, regards, Marco.

Hi, I configured Buddy Watch function on my Polycom IP 601. It works well, until I make a reload of Asterisk. After reload, if I give the "show hints" command in Asterisk's CLI, it says that there are no watcher for the extensions that I configured. Before the reload in the CLI appears: -= Registered Asterisk Dial Plan Hints =- 3002 : SIP/3002 State:

HI I have 15 vm (libvirt + kvm) with a network brigde and ip address like 192.168.0.1 VM1 192.168.0.2 VM2 192.168.0.3 VM3 ..... The vm are use by other people and I want to be sure that if someone change the ip of VM3 to 192.168.0.1 it will filter it network traffic for him and can't flood the VM1 What the best way to achieve this Hope I m clear enough

I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if

Hello! I have 2 table: users and buddies User: id, name, ... Buddy: id, user_id, user_buddy_id, ... So if you have 2 users 1,jack and 2,fred and fred is a buddy of jack, there is a Buddy object: id=1, user_id=1, user_buddy_id=2 I can declare only one belongs_to in Buddy and one has_many in User. And there is conflict if I had the second one (the first one is discarded) class User has_many

Hi. Pardon me if this is not the right place to ask. I'm new to Asterisk. I was wondering if I could use IXAy to forward my call via the internet to my destination, something of similar function to SIPURA 3000? Thank you.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, my goal is to get either blindwrite suite (v4 or 5), isobuster or any other app running that is able to create an iso of my playstation CDs. I'm using a 2.6.11 kernel and thus cannot use the ide-scsi emulation module to access my ide cd-rw. Adaptech's aspichk tool reports no aspi layer present. As I understand this is necessary for wine

There seems to be some hostility to the idea of this being a GNOME or Evolution problem: Bug 542280 ? Refuses to report bugs because gnome 2.16.x is too old View Bug Activity Product: bug-buddy Component: general Version: 2.16.x Status: RESOLVED Resolution: WONTFIX Opened by Mark Hull-Richter (reporter, points: 3) 2008-07-09 23:40 UTC [reply] On CentOS 5.2 (and 5.1 and 5.0, and Red Hat

Hi I have 1 customer who is having issues with pop3 on FC3 telnet mail.hisdomain.com 110 +OK dovecot ready. user beachy -ERR Unknown command. user beachy +OK pass letmein +OK Logged in. list -ERR No INBOX for user. pass & username fake for this example! Mark

Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the

Hi all, Been reading about chan_motif / chan_xmpp in the wiki's for 1.8, 10 and 11 version of asterisk. In each example i got the impression that the asterisk server is registering on a XMPP server as a single user with the credentials as specified in jabber.conf. Instead of a single xmpp-user, could that also be multiple users? For instance, for each sip-user an xmpp-user? When i skim

On Mon, Apr 04, 2016 at 04:46:31PM +0200, Vlastimil Babka wrote: > On 04/04/2016 06:45 AM, Naoya Horiguchi wrote: > > On Mon, Apr 04, 2016 at 10:39:17AM +0900, Minchan Kim wrote: ... > >>> > >>> Also (but not your fault) the put_page() preceding > >>> test_set_page_hwpoison(page)) IMHO deserves a comment saying which > >>> pin we are

On Mon, Apr 04, 2016 at 04:46:31PM +0200, Vlastimil Babka wrote: > On 04/04/2016 06:45 AM, Naoya Horiguchi wrote: > > On Mon, Apr 04, 2016 at 10:39:17AM +0900, Minchan Kim wrote: ... > >>> > >>> Also (but not your fault) the put_page() preceding > >>> test_set_page_hwpoison(page)) IMHO deserves a comment saying which > >>> pin we are

On Fri 18-10-19 10:50:24, David Hildenbrand wrote: > On 18.10.19 10:15, Michal Hocko wrote: > > On Wed 16-10-19 16:14:52, David Hildenbrand wrote: > > > On 16.10.19 16:03, Michal Hocko wrote: > > [...] > > > > But why cannot you keep the reference count at 1 (do get_page when > > > > offlining the page)? In other words as long as the driver knows the