similar to: Digest Username/auth name mismatch‏

Hi I want to know how to solve below an error case. Uac cant's change username of from and digest header. I tried to put aaaa at 192.168.0.1 on username of sip.conf.but same error returned. [Aug 20 14:40:12] WARNING[29315]: chan_sip.c:11806 check_auth: username mismatch, have <aaaa>, digest has aaaa at 192.168.0.1 [Aug 20 14:40:12] NOTICE[29315]: chan_sip.c:20479

Hi All; I am looking for a good Outbound Dialer and to be practical with possibility to do modification on it, the outbound dialer should send the calls to the agent when the agent is logged in as long the agent is belong to the queue (or let us say the skill group of this campaign). Any one can guide me? If I can build this using the AMI, so I appreciate if anyone did it before me so I can use

Hi all, we have a problem with a reinvite sent by our SIP provider to change audio codec due to the recognition of a fax tone. After that the SIP call session has been established (INVITE and 200 OK) we have the following codec situation: UAC ASTERISK UAS | ASTERISK UAC PROVIDER g711 <----------------------> g711

i have 2 asterisk servers one on CentOS and one on Fedora , i configured IAX trunking between the 2 servers so that i dial -say from a sip extension 2000 on fedora server to a sip extension 3000 on CentOS server the call seems to be established but hangup automatically after very short time and here is the iax2 set debug command result on centos server and also my iax.conf and extension.conf and

Hello, When dialling a SIP phone which is already in a call the caller hears a "regular" ringing tone and does not know that the called party is engaged in another call. Is there a supported way inside SIP to tell the calling party to play a stuttered ringing tone? Thanks! __Yehavi:

I need to do some testing, but before to hit by head on a known wall, i ask here. My AD domain get used (via PAM/Winbind) to give access to some other dervice, most notably here dovecot. When password expire (or users change it) the MUA try the old password some times, then ask for a new password; users cleraly get scared, press randomly 'OK' or 'Cancel', but if they press 2-3

Mandi! Rowland penny via samba In chel di` si favelave... > yes, Provided you use the right attribute to search on ;-) Ah! ;-) Just i'm here, i test three condition in account flags, eg: UAC=$(ldbsearch ${LDB_OPTS} -b "${BASEDN}" "(&(objectClass=user)(sAMAccountName=$1))" userAccountControl | grep "^userAccountControl: " | cut -d ' ' -f 2-)

I have a Steam version of DAO + Awakening and have had no troubles with it up until now when I've bought some DLCs from official bioware store. For any DLC I bought I'm getting "Unable to load area" error right after character creation or importing (or, in case of "Leliana's song" --- right after "Play" is pressed in "Other campaigns" menu). Its

I hopefully cleared all SAMBA files and set up a fresh ADC using: samba-tool domain provision --use-rfc2307 --domain=UAC --realm=UAC.MGR --server-role=dc --dns-backend=SAMBA_INTERNAL --targetdir=/srv/files --adminpass="secret" --option="dns forwarder=172.16.6.11" The provisioning seemed okay, i.e. nothing hints at any errors and I see a DOMAIN SID as the final entry as

Hello! I'm facing ReInvites as caller from UAS despite configured session-timers=refuse (which can be seen in the SIP trace) always after 900s. (The behavior is the same if session-timers is set to accept). This just happens with one provider (German Telekom to callee at kabelbw). - The incoming ReInvite is answered immediately by asterisk (Status 100 / Status 200 - 0.02s). Media stream

Is it possible to program the multi-purpose keys on a GXP-2000 remotely via a TFTP configuration file? If so, what are the parameters to put in the configuration file? Thanks, Daniel

Hi, ? I tried to use the syslinux.exe (version 4.04)?to make my usb drive bootable. When i run the syslinux.exe with the respective options I get this error ? "Accessing physical drive: The request is not supported. Did not successfully update the MBR; continuing..." ? I have administrator privilege and I started cmd prompt with "Run as Administrator" option too. I tried the

On Sat, 2014-01-11 at 06:31 -0500, Gene Cumm wrote: > On Sat, Jan 11, 2014 at 12:12 AM, Ronald F. Guilmette > <rfg at tristatelogic.com> wrote: > > > Unfortunately, regardless of whether I perform Step #4 (i.e. running > > the "makeboot.bat" script) while logged in as a user with Admin privs, > > > Accessing physical drive: Access is denied.

We have Samba setup for our shared drive. I have pasted the smb.conf file below. Everything is working well accept when we try and run an EXE file using Windows Vista. When we run an EXE file it first ask for UAC control then it pops up the username and password prompt. You must then type your username and password in again before it will run. I think the issues is that UAC is now running the

Hi all my samba version is 4.12.5 and when a sql server windows machine join the domain, It shows error in samba : Failed to modify SPNs on CN=SEC-CON03,CN=Computers,DC=domain,DC=com: acl: spn validation failed for spn[E3514235-4B06-11D1-AB04-00C04FC2DCD2-ADAM/SEC-CON03:389] uac[0x1000] account[SEC-CON03$] hostname[SEC-Con03.domain.com] nbname[DOMAIN] ntds[(null)] forest[domain.com]

I'm running Icecast in a Windows 32 bit environment. The icecast.xml path points to C:\Program Files\Icecast\log for <logdir>. <logging><loglevel> is set to 3. However, no logs are created at that location. :( Thanks for any leads. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, is it possible to use Asteriks for translating SIP to H323 and vice versa? I am looking to implement the following Setup SIP UAC <-> SIP-Server <-> SIP/H323 Proxy <-> H323 Server <-> H323 UAC Basicly i want SIP fones to talk to H323 fones and and SIP Fones to access PSTN Gateway(s) in a H323 network. Anyone got something similiar running? Any ideas? best regards,

Hello, I have such a problem. I have an * configured as a peer connected to the gateway to PSTN. While calling to the switched off cell phone, the gateway sends to the * the SIP message 180 with the SDP part, and also a lot of rtp packets containing the operator's in band info. But * forwards the 180 to the UAC without the sdp part and also without the rtp stream. Is there any way, how

Hai, small note here.. > > Setting up the Admin user is interesting. I will try that. > But I could as well add myself to the domain admins group. > The name is arbitrary. Yes the name is but now you are working with Admin rights. Never ever do that. Please dont, use the UAC. It's so easy to get infected if you work as admin so dont.. Personaly here, at the office, i

>> The call that gets dropped had a retransmission of INVITE from UAC >> to UAS (and therefore retransmission of 200 OK from UAS to UAC). >> There is nothing wrong with the re-transmission as such, but I >> noticed a potential bug in Asterisk in the way it responds to an >> INVITE retransmission. Asterisk is bumping up the session version >> number in