similar to: callerid overwrite

Hello, fresh install of Asterisk 13.11.2, client unable to register. For now I have IPtables disabled, also selinux is disabled [1006] type=friend username=1006 secret=mysecret context=sip-phone call-limit=1 callerid="iuser" <1006> disallow=all host=dynamic allow=all any ideas? Thanks, Motty -------------- next part -------------- An HTML attachment was

Hello All, one of the extensions fall into a loop, I don't know how to hangup that channel -- Executing [i at autoatten:2] Goto("Local/100 at sipphones-000001b2;2", "s,2") in new stack -- Goto (autoatten,s,2) -- Sent into invalid extension 's' in context 'autoatten' on Local/200 at sipphones-000001b2;2 -- Executing [i at autoatten:1]

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

Hello, I want to share mailbox between two extensions Ext. 101 Ext. 102 I want the messages to go to mailbox 101, when when checked mailbox from extension 102 to be able to clear the bliking red light. here is extensions.conf exten => 102,hint,SIP/${EXTEN} exten => 102,1,Dial(SIP/101&SIP/102,20,t) exten => 102,2,Voicemail(101,u) exten => 102,102,Voicemail(101,b) exten =>

Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However, every 4 or 5 times, we get an error back from the provider that says "The number you have dialed.....

Hi... I've been fighting this for a while now, trying clean builds of Asterisk 1.14.18, 1.14.19rc3, and then 1.6 Beta 6 today. No workee. :-( Here's the results for various calls made off-hook (push the blue Speakerphone button on the Polycom 430): 988852700 - Phone waits for me to either hit the soft-key "Send" or "EndCall". If I hit "Send",

Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104 at sipphones:1] MeetMe("SIP/101-00000813", "104") in new stack == Parsing

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

Follow the answers: > Yes wbinfo shows the user but does 'getent passwd iuser' show anything ? # wbinfo -i iuser iuser:*:4294967295:4294967295:iuser:/home/DOMAIN/iuser:/bin/false # getent passwd iuser iuser:*:4294967295:4294967295:iuser:/home/DOMAIN/iuser:/bin/false # id iuser id: iuser: no such user smb.conf file server: # Global parameters [global] netbios name =

> Hmm, the numbers seem extremely large, did you set this number in the > users 'uidnumber' attribute in AD ? How do I do this uidNumber configuration? I'm running all services: smbd, nmbd and winbind It's hard to run the file server as a domain member. When was a file server with DC was much more easy.

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

Hello Victor, I did set debug on, but I don?t see any errors. I did tcpdump, client is trying to register: here is the header of a udp packet User Datagram Protocol, Src Port: 55300, Dst Port: 5060 Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:pbx.mydomain.com:5060 SIP/2.0 Method: REGISTER Request-URI: sip:pbx.mydomain.com:5060 [Resent

I currently have two fax machines on my system. Both of them seem to send and receive very slowly. My end users are complaining; saying it was faster before we moved to * (Straight Analog Lines) Any help would be great. PS: I already have the d option on the Dial line. Both fax machines are in their own context: [faxes] exten => _9NXXXXXX,1,StripMSD,1 exten =>

On 10/28/2015 06:37 PM, Pete Mundy wrote: > Hi Motty, > > Isn't the whole point of the nonce in a SIP registration to ensure the > secret doesn't go on the wire in plain-text? Is this not enough, or > are you looking to hide the username too? > > (if so, fair 'nuf, just wondering why :) > > Pete > > Ps, if so then I think TLS is the missing part of

Hello, I would like to add area code if clients dial 7 digits, it that possible? currently clients dial prefix 9 plus local number, however my SIP provider is requiring to dial 10 digits. is it possible to add area code? Thanks, Motty

Hello, i'm looking for suggestions to monitor Asterisk Server? I installed Nagios but no success, I do prefer not to install any web server on the server running Asterisk. Thanks in advance. -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130509/24815a66/attachment.htm>

Hello, I hope to be in the right list. my question is the following. I can''t seem to get multipathing configure on XCP 1.6. In XenCenter I was able to enable "Multipath" however, in the LUN under general tab I see " multipath 1 of 1 active", the the CLI I do the following command xe --m node session it list only one session. Can someone help me please? Thanks in

Hi list. I've googled around for this, and so far have come up short. I'm attempting to set the outbound callerid when making a call using our PRI. I've talked with ATT (the providor) and they have said the screening table has been disabled, so anything my phone system sends for callerid should be passing along. However, when I make a call, the primary number assigned on our PRI is

Hello, I am searching for a solution to encrypt authentication from Asterisk server to clients. Searching srtp seem to encrypt traffic, I just want client authentication with encryption. Can someone point to the right direction? has anybody used ZRTP? experience with ZRTP? Thanks, _motty

Dear Rowland, This Samba 4 domain was not provisioned from scratch. Here in the company we had a DC Windows 2008. The Samba was provided to secondary DC. Then, the primary DC remains Windows, but will be removed this weekend. Samba DC will be the primary DC. In the file server file krb5.conf, I informed the KDC servers pointing to the Samba DC. Follows the smb.conf my DC Samba 4: # Global