similar to: Asterisk 1.6 - Problems with SIP/REFER

Hi everyone, Sorry if I'm repeating the e-mail, but I'm having problems with the list. I'm currently trying to enable call transfer to different domains in asterisk box (Asterisk 1.2.13 running on Debian etch). I have a configuration that requires me to transfer call to separate domains like ext at 10.10.10.10:5050. My calls come from a R2 channels in a board installed in the machine.

Hi, I am using Asterisk 1.6.2.11 to test 3rd party Voice Message server (VMS), currently when VMS send NOTIFY message (standalone NOTIFY, no previous SUBSCRIBE for the dialog for SIP), asterisk responded with 489 Bad Event, in the debug log it indicates as the following: [Feb 4 13:27:06] DEBUG[8353] chan_sip.c: Invalid SIP message - rejected , no callid, len 771 I have googled around

I am trying to debug a wine crash I am having when using GDI rendering for Starcraft BNet (Bug 5253). This bug does not happen in OpenGL mode, but my purpose is to understand the source of the bug and not to workaround it. (Bug link: http://bugs.winehq.org/show_bug.cgi?id=5253) The crash is in some unrolled memcpy loop that writes to screen memory, and as the bug description says, crashes after

Hi, I've recently discovered a scenario that causes asterisk to send SIP messages with the Request URI missing and the TO URI missing. It happens when a call goes out over a Zap channel from an internal SIP phone. When the internal SIP phone initiates a transfer to another SIP phone the transfer takes place but the NOTIFY and BYE message sent by asterisk to the first SIP phone are missing

Hi We've been running asterisk 1.4.17 (deb package) in a production environment for some while now and are finally taken the plunge to update to 1.4.38 (Ubuntu servers). All of this is using the RealTime Architecture I have upgraded the asterisk version in one of our test environments and blind transferring seems to have suddenly stopped working. It was working fine under 1.4.17 So, call

Hi, A account created with samba3/ldap (created before 2014-02-20): SID: S-1-5-21-XXXXXXXXXX-XXXXXXXXX-XXXXXXXXXX-3216 UidNumber : 1108 A account created with Users and computers (samba 4 AD DC) SID: S-1-5-21-XXXXXXXXXX-XXXXXXXXX-XXXXXXXXXX-5878 uidNumber : 10023 My actual config (in file-server) : idmap config XXXXXX:backend = ad idmap config XXXXXX:schema_mode = rfc2307 idmap config

Hi all, I have the following 7 x 7 matrix. ?I am trying to figure out how to label the columns to something more descriptive other than [,1], [,2], etc. I have tried the c(x,y,z,) function, but I get a error returned stating that my vectors need to be the same length. Do I need to convert this to something else such as a list and then repack it? Thanks, Kindra ?? ? ? ? Volume Time ? ? ? ?[,1]

Hi, I am running a Asterisk 1.6 box in our Data Centre, and have a number of users connecting to that box, as their PBX. Calls in and out work fine, as does voicemail. The PBX at the Data Centre has an External IP, Nat?d to it by the firewall, and the relevant ports are open. The Office users have a dedicated internet connection for the phone lines, and calls are seen to traverse this

I am having a problem with SJphone registration, having read the list and wathced it for a while for similar problems. I just can't seem to figure out the problem. I tryed to follow a tutorial from http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone, but in SJphone (SIP tab), I can't find the following setting. Use local outbound proxy - checked. Proxy IP Address:

Hullo :) I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using CAPI for ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from sipgate.co.uk to any other extension. My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind transfer, simply dial the number you want to transfer to, and press 'FWD'... This is what

Hey all, I noticed that my puppet server running CentOS 6.5 was acting a little pokey. So I logged in and did what well just about anyone would've done. And ran the uptime command to have a look at the load. And it was astonishingly high! [root at puppet:~] #uptime 21:28:01 up 1:26, 3 users, load average: 107.37, 72.06, 75.52 So then I had a look at top and saw a LOT of processes

TEQL and Subnet problem I have a network topology shown below, and I am trying to use TEQL. My problem is: When I ping to P3.teql0 from P2 ("[P2]# ping 16.119.144.66"), the traffic can never go from P2.eth1, and all traffic only goes to P1.eth0. What P2.eth1 (16.119.144.33) did is broadcasting an ARP asking for the MAC address of 16.119.144.66, although I have specified the route to

Hi Everyone, I can connect the Jabra GN2124 + GN2100 (smart cord) to the Aastra 53i receiver port and I get a tone. But when I connect it to the headset port there is no tone. I am running firmware 2.4 and I can't seem to find that DHSG, EHS or whatever the setting maybe called to enable to get this headset work with the phone. Can anyone quickly tell me where the audio options are on this

On Mon, 29 Jan 2024 22:07:36 +0100 "Dr. Nicola Mingotti" <nmingotti at gmail.com> wrote: > Done, it says what I would expect, the Domain Controller name is DC1 > > foo at dc1> sudo samba-tool user show nicola > ERROR: Failed to get password for user 'nicola': Unable to find user > "nicola" > > foo at dc1> sudo samba-tool user show

We recently upgraded from Dovecot 1.2.15 (on Debian squeeze) to Dovecot 2.1.7 (on Debian wheezy/stable). For now, we continue to use FTS with Squat indexes. We'd like to eventually switch to Lucene but it's not yet supported in Debian stable. Since upgrading to 2.1.7, we've seen some problems with IMAP search results when an FTS Squat index is present (i.e., when the

Hello r-devel, The attached Code page 437-encoded file contains 245 characters (including the final newline), but readLines only reads 242 of them: > test_text <- readLines(file('437__characters.txt', encoding='437')) Warning message: In readLines(file("437__characters.txt", : incomplete final line found on '437__characters.txt' > test_text [1]

Hi, What you're trying to do is mixing RID and rfc2307. This is not possible. I've the same kind of issue here (Samba 3 migrated DC with samba unix users created in the same range as regular unix users), but still use rfc2307 so I can renumber users one by one as follow : * Save old uid (1000-2000 range) * Give a new one (10000+ range) * Launch a command like (multiple -e are

Hello my friends... Currently we are using the following firmware versions on ours aastra 55i: Firmware Information Attribute Value Firmware Version 2.1.0.2145 Firmware Release Code SIP Boot Version 2.0.1.1055 Date/Time Jun 20 2007 06:20:29 Can we make a firmware upgrade to the latest one: 6755i (55i) SIP, V2.5.3.18, January 2010 , English , ZIP , 2,849 KB on the site:

Could anyone help me set up Asterisk in such a way that it makes SIP-SIP transfers using the REFER / NOTIFY method according to RFC-3515 ? SCANARIO: - Asterisk registers with PSTN<->SIP VoIP provider "V" (Vonage) as a friend - Asterisk is located in Europe, Vonage in located US. - Asterisk acts as an autoattendant located in Europe. - Asterisk answers and incoming call from

I've completed my mailbox rebuild - theoretically I should be free of corruption. I used dsync to export from mdbox to maildir (so should be clean) then used a virtual machine with Dovecot to import back to mdbox in another location. So...theoretically I should be free of all corruption now... Running an fts update - "doveadm search text -u user at domain.com xyzzyx" works on