search for: company1

...isplay of which line the call came in on. For example, the receptionist answers calls for 8 different companies and would like the phone to display the company name that she should announce to the caller. Here is a more complete output of an incoming call. I've changed the SIP numbers to "Company1', etc, to hide the numbers. Connected to Asterisk 10.12.4 currently running on asterisk (pid = 32267) > Verbosity is at least 12 > asterisk*CLI> > asterisk*CLI> > asterisk*CLI> > == Using SIP RTP CoS mark 5 > -- Executing [s at incoming:1] *Set*("*SIP/Comp...

...NS and DomainDNS FSMO roles from Windows DC to Samba (other roles transferred successfully). (dc01 - is a Windows DC, linux01 - Samba DC) [root at linux01 ~]# samba-tool fsmo show SchemaMasterRole owner: CN=NTDS Settings,CN=LINUX01,CN=Servers,CN=Default-First-Site-Name,CN=Sites,CN=Configuration,DC=company1,DC=dd InfrastructureMasterRole owner: CN=NTDS Settings,CN=LINUX01,CN=Servers,CN=Default-First-Site-Name,CN=Sites,CN=Configuration,DC=company1,DC=dd RidAllocationMasterRole owner: CN=NTDS Settings,CN=LINUX01,CN=Servers,CN=Default-First-Site-Name,CN=Sites,CN=Configuration,DC=company1,DC=dd PdcEmulati...

...ugh what line the call came and a beautiful display of this information. 1. definition through what line the call came. If the username and password for inbound and outbound registration the same, then try the following: a) delete "register" lines. b) add option "callbackextension=Company1" to Company1 friend section.. And in others with their names too. or you can change "/s" to "/Company1" in register line. 2. beautiful display of this information a) add option "setvar=fromCompany=Company1" to Company1 friend section.. b) In dialplan add Set(CAL...

...ile: exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one exten => s,2,Set(pseudodid=${SIP_HEADER(To)}) exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)}) exten => s,4,Set(pseudodid=${CUT(pseudodid,:,2)}) exten => s,5,Set(callersname=${IF($[ ${pseudodid} = 081...]?Company1:${callersname})}) exten => s,6,Set(callersname=${IF($[ ${pseudodid} = 082...]?Company2:${callersname})}) exten => s,7,Set(callersname=${IF($[ ${pseudodid} = 083...]?Company3:${callersname})}) exten => s,8,Set(callersname=${IF($[ ${pseudodid} = 084...]?Company4:${callersname})}) exten =>...

...MO roles from Windows DC to Samba (other roles transferred successfully). > (dc01 - is a Windows DC, linux01 - Samba DC) > > [root at linux01 ~]# samba-tool fsmo show > SchemaMasterRole owner: CN=NTDS Settings,CN=LINUX01,CN=Servers,CN=Default-First-Site-Name,CN=Sites,CN=Configuration,DC=company1,DC=dd > InfrastructureMasterRole owner: CN=NTDS Settings,CN=LINUX01,CN=Servers,CN=Default-First-Site-Name,CN=Sites,CN=Configuration,DC=company1,DC=dd > RidAllocationMasterRole owner: CN=NTDS Settings,CN=LINUX01,CN=Servers,CN=Default-First-Site-Name,CN=Sites,CN=Configuration,DC=company1,DC=dd...

...ve the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses. [a00090101] type=friend context=Company1 username=a00090101 ;secret=180 ;insecure=very host=dynamic mailbox=company1@vmusers deny=0.0.0.0/0.0.0.0 permit=192.168.10.4/255.255.255.0 permit=192.168.10.5/255.255.255.0 accountcode=a00090101 subscribecontext=status ;qualify=yes canreinvite=yes I'm trying to configure Asterisk to accept con...

Hello all, I have an Asterisk server (Asterisk 10.12.4) with multiple sip accounts with the same service provides. We have 8 phone numbers in total. Incoming calls from the public are all correctly directed to appropriate office handsets. However, the display on the reception phone (the only one i care about) is always showing the same "SIP/Account1_0843214321" rather than the account

...te=yes is enabled. No dial command has 't' option. However, it seems that Asterisk still stay in the media path and bridge the 2 end points. Am I missing something??? sip*CLI> show channels Channel (Context Extension Pri ) State Appl. Data SIP/22225001-c60b (company1 1 ) Up Bridged Call SIP/1234-faf1 SIP/1234-faf1 (company1 5001 1 ) Up Dial SIP/22225001|20|r 2 active channel(s) sip*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.1.101 2222500...

Hi everyone! I have very simple question, how to limit SIP phone user making calls to for example longdistant calls? Maybe: Put in his context in sip.conf context which don't provide possibility to make such calls? Is it correct? thanks for any help, regards, Corvin

Hypothetical question.. Lets say there is a situation where you are using the highest compression codecs for all extensions (I guess that would be G.729) and the load on a single server is overpowering the most powerful single processor(lets say SMP is not an option).. So two or more servers are required.. Or The situation is that you need fault tolerance so want to have two Asterisk

...fine. Now the notification emails of course should be customized per wholesale customer. I added a 'mandate' table to the database and get this field by an AGI script before calling VoiceMail to get the correct language and context name for this particular mailbox. Let's call them company1 and company2 Then I do: exten => s,n,AGI(getmandate.agi) exten => s,n,Set(CHANNEL(language)=${MANDATELANG}) exten => s,n,VoiceMail(0${CALLERID(rdnis):2}@${MANDATE},u) in voicemail.conf I have: [company1] serveremail=voicemail at company1.example.com tz=european emailsubject=[Customer 1...

...> |----eth2(Company2) 80kbps (203.145.134.116/255.255.255.252) > & > (192.168.100.0/255.255.255.0) > |----eth3(Company1) 120kbps(192.168.200.0/255.255.255.0) > [snip] > _______________________________________________ LARTC mailing list / LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/mailman/listinfo/lartc HOWTO: http://lartc.org/

I would like to setup a system with two tennants with two seperate DIDs through one IAX provider account. Is it possible to route the calls into different contexts based on the DID dialed? I have searched and found nothing. I do not see anywhere in the console that says what DID was dialed so I am thinking two seperate accounts are needed to make this work. Can anyone confirm? Thanks

...y simple question, how to limit SIP phone user making > > calls to for example longdistant calls? > > This is how I do it - Thank you very much to all of you. I have one more question which troubles me. We have scenario: (only SIP is considered now) Subscriber A registered in Asterisk company1.org (eg. A@company1.org) Subscriber B registred in Asterisk company2.org (B@company2.org) How it is possible to make connection between those. Their extensions (called party) (maybe I don't understand term extension correctly) is not definied in corresponding extensions.conf. Extension B is no...

...Samba4 so I very much hope that somebody on this list can help me out. Basically I try to authenticate users through the Outlook autoconfigurator using RPC-OVER-HTTP to a samba server. The problem is that in Samba4/LDAP I cannot have users with email address in their name so instead of: user at company1.com I can only create user_company1.com I don't detail it more because on the top link I have explained everything with pictures. So what I would like to know is how to REBIND the incoming user auth request into a new format if this is possible. auth_check_password_send: Checking passwor...

Good day all I want to put the openline4 card into a box that will support 3 different companies I read the caller ID id fixed but now HOW DO I: If a call come in for 12345 it plays company 1's welcome message If a call come in for 98765 it plays company 2's welcome message ens.. Does This make sense Thanks Altus

...Samba4 so I very much hope that somebody on this list can help me out. Basically I try to authenticate users through the Outlook autoconfigurator using RPC-OVER-HTTP to a samba server. The problem is that in Samba4/LDAP I cannot have users with email address in their name so instead of: user at company1.com I can only create user_company1.com I don't detail it more because on the top link I have explained everything with pictures. So what I would like to know is how to REBIND the incoming user auth request into a new format if this is possible. auth_check_password_send: Checking passwor...

All, when I do show channels: Channel Location State Application(Data) SIP/201-e478 (None) Up Bridged Call(IAX2/muncie_to_ge IAX2/muncie_to_georg s@default:7 Up Dial(SIP/201|20) I am getting TRUNCATED call information.... the IAX2/muncie_to_ge is truncated. How do I get the need call information to transfer the call. Jerry

...n below (without all the shares, which are all the same). All pc's are set to be a member of "WORKGROUP". Any ideas? Thanks, Jack D. Browning jac at netNOSPAMdor dot com ----------------- begin smb.conf ----------------------- [global] workgroup = WORKGROUP server string = company1 Samba Server hosts allow = 10.0.0. 127. log file = /var/log/samba/smbd.log max log size = 50 smb ports = 139 445 encrypt passwords = yes smb passwd file = /etc/samba/smbpasswd unix password sync = Yes socket options = TCP_NODELAY SO_RCVBUF=8192 SO_SNDBUF=8192 os level = 3...

I have an * system running version 1.0.8 and it works mostly fine. I am using it as a virtual PBX and we share the box among companies. I have the calls all staying separate, we well as the companies' extensions, voicemail, etc. The only problem I'm having is with two accounts that use the same SIP termination provider. * seems to be confusing the sip contexts for the incoming calls.