similar to: sip realtime broken?

Boa tarde, Ap?s alguma experi?ncia com o Asterisk, e com muito ainda para aprender, gostaria de saber se h? algu?m nesta mailing list que pretenda criar um Asterisk Users Group para Portugal. Visto que acaba sempre por ser uma enorme aprendizagem ( valor acrescentado) a partilha de experi?ncias/problemas e solu??es nas implementa??es Asterisk. H? spre detalhes que variam entre os Telco's de

hi all i have an asterisk working and i need to add a mettme public service. for example i need to download a soft (sjphone) and without any configuration call to 509@asterisk.mydomain.com (meetme) and join a conference but when i do that i received an error saying nomber do not exist. but if i call a extension is work propperly. in the extensions.conf have exten => 411,1,Answer exten

Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. For alcatel users use asterisk lines, should dial 0 to take tone from asterisk. In default configuration in alcatel, if user dial 0 this error occour: !! Unexpected Channel selection 3 -- Extension '' in context 'default' from '' does not exist. Rejecting call on channel 0/31, span 1 In alcatel

Hello, I'm french, so excuse my poor English. I'm face to a terrible thing, with has stole a lot of my time. On the .184 machine, I've the following iax.conf : [general] rtcachefriends=yes bandwidth=high tos=reliability jitterbuffer=no autokill=yes #include "iax.voip1.conf" #include "iax.renoir.conf" The iax.voip1.conf file contains : [VOIP1] type=friend

Hi guys, I have an Alcatel 4200 with a ISDN board pluged in the asterisk server with TE110P. Input calls VOIP Proider ---> Asterisk ---> Alcatel Output Calls VOIP Proider <--- Asterisk <--- Alcatel In alcatel phones, users should dial 2 for take a line tone and can dial. At this point start my problems: 1. When users dial 2 on phone (alcatel) they don't received a dial tone,

I'm running asterisk 1.2.8. I would like PSTN inbound calls to do the following: 1-once PSTN callers enter their desired extension; they have to record their name 2-recording then announces that it is trying to locate the user 3-asterisk calls local extension and announces callers recorded name 4-local recipient user can choose to take the call, send it to voicemail or transfer it to

Hi list, I have a server running a simple hfs isdn card running with chan_misdn. the problem is, I can't get asterisk to pick up the phone, outgoing calls work fine. when running asterisk with -vvvvvvvvvvvc I get the following "log" when I try to dial the isdn server. P[ 1] * Starting Ast ctx:kpn-in dad:0594643637 oad:0878722291 with 's' extension P[ 1] MGMT: SSTATUS:

I'm interested in knowing if anyone else has worked around this issue: I have an application that needs to check the status of the calls going through Asterisk about every 5 seconds or so. I don't want to do "asterisk -rx 'show channels verbose'" at the Linux command line 12 times per minute so I am looking at the AMI. I see that there isn't a manager command

Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem

Hi all, I would like to do some QSAR analysis (quantitative structure activity relationship). I need to use some Partial Least Squares (PLS) regression, but I have not seen this option on the R-project. Is it possible to do this kind of regression on R? thank you in advance best regards, olivier [[alternate HTML version deleted]]

Is the person responsible for the FreeBSD port on this list? There might be a minor bug in a small piece of correction code issues in the last 4 days. -- Member - Liberal International This is doctor@@nl2k.ab.ca Ici doctor@@nl2k.ab.ca God,Queen and country!Never Satan President Republic!Beware AntiChrist rising! http://www.fullyfollow.me/rootnl2k Look at Psalms 14 and 53 on Atheism Time for

Hi, could anybody suggest a usable doorphone and magnetic door opener "hardphone" system for me, please? Of course should be connectable to asterisk. I am in the EU, should be available here. thank you, Csaba

Hi everybody! Is there any package or functions to make Partial Least Squares analysis with R? Thanks a lot Luis -------------------------------------------- sapo.pt/kitadsl -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send "info", "help", or

Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj

The SIP channels have a "call-limit" parameter (which is badly documented and I haven't tested yet) How can I have the same behaviour for IAX channels? I can't see anything related to it. Ah, I'm using Asterisk 1.2.13... maybe there is something in the 1.4 versions... but I can't change to 1.4 right now because of MFC/R2 BarZ

Does anyone know where to find an example or able to provide an example of how to do the following: When asterisk answers a call... Ask for number to dial...then dial that number? I am basically dialing into the asterisk box and then wanting it to take the digits I enter and dial them on an outbound zap trunk... I basically am just not sure how to have asterisk accept the digits and then use

Hi all, We have Xeon-based system with only 1 (hyperthreaded) CPU (in a HP DL360). We are seeing high load on multiple meetme session as well as g729 transcoding. My question is will putting an extra CPU help or does Asterisk just run on a single CPU. -------------- next part -------------- An HTML attachment was scrubbed... URL:

---------- Forwarded message ---------- From: Marco Mouta <marco.mouta@gmail.com> Date: Feb 15, 2006 1:58 PM Subject: Which ATA device do you recommend? To: asterisk-users-request@lists.digium.com Hello, I'm developing a Voip Solution for a client, which ATA SIP do you recommend? there are some ATA devices fully tested with Asterisk? I hope that Asterisk experient users could give me

Hi list, I'm with a little problem on codec negotiation between a cisco827 and asterisk. My sip.conf is like that: [general] port = 5060 bindaddr = 0.0.0.0 context = default amaflags = default allow=g729 allow=gsm allow=alaw allow=ulaw ;disallow=all and cisco like that: dial-peer voice 6 voip destination-pattern 0T session protocol sipv2 session target ipv4:<asterisk-ip>

S?rgio Henrique <sermac at gmail.com> quatschte am Mon, Feb 25, 2013 at 04:26:30PM +0000: > Solved. > > I have sucessfully migrated a windows 2008R2 domain to samba4 and then > create a new samba domain as a replica. > > A lot of steps i had to introduce. Hi S?rgio, > 1- Working on DNS > add samba dc to forest and domain dns _ldap values > change DNS SOA to