similar to: Accountcode and SIP Peers Part 2

Hi to all, to manage properly a call center for multiple companies is possible to let the X-lite/X-Pro softphone to display the number or context called from PSTN to let operator answer with the correct name of the company?? I explain better. If a call come from PSTN to Number A for company A i want the operator recognize it and answer "Good Morning, I'm Operator of company A"

Hi, i read the * support ringing multiple devices at the same time, i inserted this line on my configuration on default context: exten => s,1,Dial(SIP/260&SIP/261&SIP/262&SIP/263|30) exten => s,2,Voicemail,u260 exten => s,3,Hangup And i have both 4 clients in sip.conf . The problem is that if i call it fall immediately in the Voicemail if the client 260 is not registered .

Hi to all, i saw that in chan_sip there is the possibility to let the * to take the number from the Remote-Party-ID header field on incoming calls from gateway. What about to let the * to generate the Remote-Party-ID on outgoing calls? this is is useful for us to let the users to have their outgoing number hidden but let our switch to get the correct record for accounting. I think that If i hide

Hi to all, can you suggest to me the best way to avoid problems in the CDRs for anonymous sip calls? I have some peoples that set Send Anonymous : Yes in their Grandstream phones and i don't receive the username as phone number that i use to make billing. It is empty. The only place where there is the phone number is in the peer name where it write the name of the peer that in this case is

Hi to all, i already searched the archive without finding a solution to my problem. I have asterisk installation 1.2.18 to support multiple virtiual PBXs. I use SIP peer in the format <ID>-<EXT> to let every virtual PBX to share the same numbers of EXT. Ex. (PBX ID 10 Extensions) 10-101 10-102 10-103 (PBX ID 20 Extensions) 20-101 20-102 20-103 I use some rules in the dialplan to

Hi to all, we have a community of people on an * box that use SIP softphones to talk each other. Can you suggest me the quickest and simple way to let someone know who is online without have to call one by one the persons to look if they are present or not?? Something the user list in Microsoft Messenger. I was thinking on some sort of web page that can check the registration of the sip

Hi to all, following the suggestion of Matteo Brancaleoni to solve my problem of anonymous call in CDRS i implemented the accountcode in any peer that i have in the sip.conf so i can use the SetCallerID to accountcode to identify the caller. When i make the sip show users the accountcode is correctly displayed but when i try to access the variable ${ACCOUNTCODE} from the extensions.conf to

Hi, i'm using * with SER and a cisco 3725 as Gateway. I noticed that the reinvite is not working if i use SER and if i don't use IT (*---->Gateway) the reivite works so the * server is able to let the RTP direct from gateway to SIP Clients. Do you know in which way can i let it work with the SER too. Becouse i need SER to manage other VOIP communities but if i'm not able to use

Hi, I use realtime and I found that changing accountcode needed to restart asterisk to activate that code and shown in CDR. Does it has a way to update accountcode without restart asterisk? ango

I've just found out that CDR(accountcode) variable can only be 20 characters long, doesn't matter what size the MySQL column has for it. I need to increase it to at least 30 characters. Any idea how this can be accomplished? -- Zeeshan A Zakaria -------------- next part -------------- An HTML attachment was scrubbed... URL:

Greetings list, Whilst running through a routine check of some CDRs, I've noticed that the originating channel's accountcode isn't preserved on creating a local channel. For example, if we start with: exten => 123,1,Set(CDR(accountcode)=foo) exten => 123,n,Queue(bar,nrtw,,,) And the queue 'bar' is defined as follows: [bar] member => Local/456 at outbound member

Hi All, I'm running 1.2.9.1 and have a sip user setup with accountcode=4444 in the context. lab1*CLI> sip show peer 1234 * Name : 1234 Secret : <Set> MD5Secret : <Not set> Context : sip1004 Subscr.Cont. : <Not set> Language : Accountcode : 4444 AMA flags : Unknown CallingPres : Presentation Allowed, Not Screened Callgroup

I have a sip.conf entry for a customer's PBX (IP based authentication) that reads: [customer] type=friend context=customer host=x.x.x.x accountcode=10000 disallow=all allow=g729 When the customer makes a call to my * server, * recognizes the peer correctly. However, for some reason, the AccountCode is blank. I have a NoOp(${ACCOUNTCODE}) and the CLI shows: -- Executing

I've setup cdr_mysql and am using AGI to authenticate users based on the called-from # (callerid), use the AGI perl module. Looking at the info stored in the caller detail, I see a field called "accountcode", is it possible for me to set this field in AGI? I'd like to tie it to a username, that I pull during my SQL authentication, so I can search the cdr table based on a

Hi, I`m occassionally seeing CDR(accountcode)'s value empty at a place in my diaplan where it was filled with some value a few lines before, with nothing else having changed it. It`s giving me headaches (as I rely on it for MySQL queries). Anything I can do? Mick -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I have about 10 SIP phones for different users defined in sip.conf, each with their own accountcode= entry. There is a global setting in sip.conf that states amaflags=documentation There are 3 IAX->PSTN gateways defined in iax.conf for outbound calls. These do not have an accountcode=, but do have amaflags=billing defined in each. The theory was that all calls should be logged, those

Hi, I want to assign different accountcodes (for billing) according to the IP address and or the H.323 name (chan_oh323). I tried in extensions.conf something like setVar(ACCOUNTCODE=userid) but in cdr I find the accountcode set in oh323.conf. Howto change it in extensions.conf? Roger.

Trying to do something like this in the sip.conf under my incoming provider profiles: setvar=CDR(accountcode)=${EXTEN} It seems to show up in the CDR but it's showing up exactly like this "${EXTEN}". Is there a way to stuff the DNIS (number dialed) into the accountcode for CDR? I have already accomplished setting on a number by number basis, I just want to do it globally for all

Hi I have two Asterisk Server: The first server "A", all phone are connected The Second server "B" only route call to a lot of SIP supplier the server A sent: ; Destination: Non connu dans le DialPlan - Apparaitra en UNKNOW dans le CDR exten => _X.,1,Set(CDR(CodeTier)=BUS-UNKNOW) exten => _X.,2,Dial(IAX2/SERVERB/${EXTEN},180,rt) exten =>

Hi All, Call comes into Asterisk Asterisk answers and Dials SIP Phone SIP phone has call forward enabled to a long distance number Asterisk receives a SIP response 302 "Moved Temporarily" back from phone Asterisk then forwards inbound call to 'Local/number@context' thanks to phone 2 problems with the CDR: 1. intermittent 'bill sec' accuracy, sometimes 0 even when the