search for: uac

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 | g729 <---------------------------> g729 rtp rtp After a while, we...

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=$1))" userAccountControl | grep "^userAccountControl: " | cut -d ' ' -f 2-) # Old 'D' flag: ((($UAC & 2) == 2)) && enabled="false" # 0x0000...

...ndard solution (profile.dap moving/deletion, AddIns.xml manual rewriting, reinstalling via daupdater.exe, verifying steam game cache integrity) had no effect so far. EA support as a last resort offered some weird solution which I could not properly try on Wine because it suggest disabling/changing UAC settings and disabling some background services. Does anyone played official DLCs and had successfully overcome "Unable to load" errors? Please, share you suggestions, any help would be very much appreciated! In case it'd help EA solution (sorry for the wall of text): > Andrew M...

...eplace: badPwdCount" >> $TMPLDIF echo "badPwdCount: 0" >> $TMPLDIF ldbmodify ${LDB_OPTS} "$TMPLDIF" > /dev/null but do that if and only if account is locked, and i test that using: user_is_locked () { local locked="false" local UAC=$(ldbsearch ${LDB_OPTS} -b "${BASEDN}" "(&(objectClass=user)(sAMAccountName=$1))" userAccountControl | grep "^userAccountControl: " | cut -d ' ' -f 2-) # Prevent error conditions on query error. if [ -z "$UAC" ]; then...

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 well as a fresh smb.conf...

The Asterisk Development Team has announced the release of Asterisk 1.8.9.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.2 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * ---

Hi gentleman I've configured SER to forward every call starting with sip uri request "1" to Asterisk. I need to configure Asterisk as a Sip UAC in order to make it call to my other SIP Provider outside my network, sending username and password for authentication. I've read at www.voip-info.org some articles but found none that could suit to my needs, but yet I've found an article which explains an implementation very similiar to...

Hi, Using openser 1.1.0-dev8 as a registrar/proxy in from of Asterisk. Recently I have been getting errors from Asterisk due to corrupted From: headers, which appear to be caused by uac_replace. Here is a section of the debug log: Mar 14 15:12:00 www1 /usr/sbin/openser[7933]: DBG:uac::restore_from_reply: removing <From: <sip:lenc_domain.com@sip.domain.com>;tag=635c3ce6 > Mar 14 15:12:00 www1 /usr/sbin/openser[7933]: DBG:uac::restore_from_reply: inserting <Fro...

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:

Hi everybody, I have a SER Server (Sip Proxy / REGISTRAR) and a Asterisk Server (PSTN and other services). I've got some clients that make calls to each other through my SER Server, that's to say, non external or international calls. I would like my clients to make external and international calls through my server but for that they must authenticate at another server to have a valid

...sip.c:20479 handle_request_register: Registration from 'aaaa <sip:aaaa at 192.168.0.1>' failed for '192.168.0.2' - Username/auth name mismatch [aaaa] type=friend username=aaaa secret=xxxx context=bbbb canreinvite=no host=dynamic disallow=allallow=ulaw The error seems that UAC set different username of digest. But UAC cannot send same username of digest and from for specification. *Digest username set aaaa at 192.168.0.1 So I want to know how to solve with Asterisk. Register From: aaaa <sip:aaaa at 192.168.0.1>;tag=644056924 To: aaaa <sip:aaaa at 192.168.0.1...

hey all, am wondering if anyone has successfuly done a SIP attended transfer using the REFER method (after an INVITE obviously) and the Replaces: header. we're writing our own SIP UAC and the asterisk code seems to support it, but we're not really sure if this is so. we plan on the following call flows: 1. incoming call from exten 1111 is sent to our UAC with Dial() 2. our UAC makes another call (via asterisk) to exten 2222 3. 2222 answers the call 4. our UAC sends REFER...

...register" command, for instance. There are also some other things that are not completely clear. That's why I wanted to ask lots of questions about it, and hopefully I'll get some good answers here. I'll focus only on having Asterisk being a SIP *client* (I believe this is called UAC [User Agent Client] in the SIP world, right?) connecting to a SIP provider (such as FWD). I'm using asterisk 1.4.6, so I would like to talk *only* about configuration on asterisk 1.4.x. * SIP channels on outgoing calls If on my sip.conf I have a section [myprovider], it always creates a new...

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 have the same restrictions as every other user. Except i can overrule it with UAC, but i never work with admin rights ever, not even at home. Greetz, Louis

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, Arnd

...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 to setup the * dialplan to translate all incoming 180 SIP messages to 183 with the SDP part and also to forward the rtp stream to the UAC?? Thanks for advices... Tomas

>> 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 vers...

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 application under Admin instead of the logged in user. So the first username and password that is cached is not...

...tered to as user) will recognize that the call should be placed on hold. A good example of what I want to achieve is presented in Section 2.1 of RFC 5359 (Session Initiation Protocol Service Examples) ( http://tools.ietf.org/html/rfc5359#section-2.1) where "Bob" would be my Asterisk (as UAC), "Alice" is the external caller and "Proxy" is the provider's SIP server. Question: ---------- Is there any way to perform this from the dialplan or by means of the manager API? Is there an application like "Hold"? Kind regards and greetings from Austria, John...