similar to: SNMP for Asterisk

Good day all How do I get my asterisk and sip to use the password.I'm using x-lite.If I use just the username and no password it still logs on? Here is my sip.conf entry? [101] type=friend callerid="Test User" <101> context = test_1 ; Default context for incoming calls username=101 secret=123456 host=dynamic dtmfmode=inband ; Choices are inband, rfc2833, or info

Goos morning all, I compiled Samba 3.0.14a with OpenLDAP 2.1.22-0 directory. I then enabled TLS between Samba and OpenLDAP. The following tests succeeded: s_server to s_client --> OK slapd to s_client --> OK slapd to OPenLDAP client commands (ldapsearch..) --> OK The problem is the following: when I start Samba (service smb start), slapd output returns: TLS trace:

Can someone who is knowledgable in the traditional telco space please give me a layman's explanation (or point me to an appropriate url) of LBO as per the zaptel configuration file? # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1)

Hi, I use Asterisk 1.0.5 and TDM04B. When an incoming call over ZAP channel hangs-up, it takes 10 seconds until Asterisk realize that. How can I shorten the time of hang-up detection? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050614/de3eec0f/attachment.htm

Hello, I patched asterisk cvs head sources with http://juraj.bednar.sk/work/software/asterisk/messaging/ and presnce patch without success. asterisk send "405 method not allowed" to sender. I use polycom ip300. Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger

Hello, I try set Ua---SER----Asterisk (voicemail/ARA) | Ua ser stable asterisk cvs head I read http://mail.iptel.org/pipermail/serusers/2005-February/015997.html to forward unavailable or busy sip agents to asterisk voicemail in failure route. How may I configure extensions.conf and ser.cfg ? I have been trying without success! Regards Harry

hello, I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. Anybody could help me to configure Asterisk in order to set instant message and presence ? I've tried with Ondo sip server it's ok ! Regards D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail

What's wrong with ARA (asterisk realtime architecture) from voip-info: Asterisk, SER and MWI http://mail.iptel.org/pipermail/serusers/2004-December/013727.html Actually I wrote a patch for this and it supports ast_data too. What you do is tell asterisk that all of your phones IP addresses are your SER machine. Then when a message gets left Asterisk sends the NOTIFY to username at

Hi, Please, what does this message mean: "getpeername failed. Error was Transport endpoint is not connected"? I get the following in my log.smbd file each time I try to log to my samba domain from a Windows XP client: [2005/08/24 09:49:09, 0] lib/util_sock.c:get_peer_addr(1150) getpeername failed. Error was Transport endpoint is not connected [2005/08/24 09:49:09, 0]

hello, I need help with ast_data I downloaded asterisk from cvs cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot co -r HEAD asterisk and the latest ast_data. When i run ./INSTALL.txt i get : serveur1:/opt/asterisk/ast_data# ./INSTALL patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c Hunk #1 succeeded at 27 with

Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600.... Is there any additional debug apart from "show hints" to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =-

Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts "403, Forbidden" . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I

Dear Asterisk community, sorry if I'm so stupid, but I couldn't register myself with Asterisk. I created the [sip-incoming] context in the sip.conf: [sip-incoming] type = peer username = elzhov port = 5062 ; my kphone listens port 5062 host = 127.0.0.1 Then run Asterisk, and checked peers that are known for Asterisk: *CLI> sip show peers Name/username

Note: forwarded message attached. __________________________________ Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html -------------- next part -------------- An embedded message was scrubbed... From: Khubeka JM <jmkhubeka@yahoo.com> Subject: JE TROUVE QUE VOUS N'ETES PAS HONETE! Date: Sun, 26 Jun 2005

Greetings, Recently there has been talk of the presence/buddy feature with asterisk and Polycom phones. I have it setup, and working as expected, however I can only get 7 buddies to appear on the screen at any given time. Has anyone gotten more than 7 buddies to appear? I'm just trying to find out if this is some polycom limitation, bug, or my error. Thanks, Matt -- Matt Gibson VOIP

Hi all, I have a problem starting the ztdummy. Here is what happens: [root@asterisk /]# modprobe ztdummy Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' 1 error(s) detected FATAL: Error running install command for ztdummy After this, ztdummy is visible with lsmod, but when I try MeetMe, I get following: == Parsing

Hi, Can anyone help me with this: I have downloaded latest stable version of Asterisk using the asterisk-update.sh script. Compilation and installation passed well. When I start Asterisk I get the following error: [pbx_realtime.so]Jan 28 09:35:08 WARNING[3253]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_realtime.so: undefined symbol: ast_load_realtime_multientry Jan 28

Hello, I wish to use asterisk as a voicemail server with ser . I want to use asterisk external configuration toHello, I wish to use asterisk as a voicemail server with ser . I want to use asterisk external configuration to manage users and storing voicemail messages according to ser database. Where can i find the schema of the SQL DB for voicemail accounts . for example in extconfig ;

Hi all, I had a chance to use some call conferences that had some very neat functionalities: - When you call you are first asked for your name - When someone joins the conference a message "<name> is now joining the conference." is played. - When someone leaves the room a message "<name> has left to conference." is played. How can I set MeetMe/Asterisk to have

Hi, I have stress tested the Asterisk Voicemail. We have encountered problem with simultaneous calls that are sent to the same mailbox. It occurred that several calls were writing to the same file. It seems that there is a synchronization issue in the Voicemail application. Did someone else find this issue? What would be the solution/workaround for it? Regards, Stojan Sljivic