similar to: Accountcode and SIP Peers

Hi, notice that i have Grandstream phones and i have the problem if i activate the Send Anonymous function on them. If i do not activate that option the ACCOUNTCODE is correctly populated. SO i think it may be a bug of asterisk. I'm using Asterisk CVS-HEAD-10/07/04-18:07:25 . Thanks, Bye, Marcello

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 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

Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN -> G729 mixing. Just SIP -> SIP using G729 for calling remote offices via VPN, but everything else use G711. -----Original Message----- From: Brancaleoni Matteo [mailto:mbrancaleoni@espia.it] Sent: Wednesday, July 16, 2003 11:32 AM To:

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'm experiencing a issue (not big, but important) I have an asterisk installation with a buch of sip phones & analog ones. I have 2 1 sip phone that's outside in the "world", and is nat'ed. I'm using g.729 with it. I wanna use g.729 only for the remote phone, and ulaw for the local ones, since they're on a lan. What happens? when I call the remote phone, g.729

I was wondering if the accountcode flag works with sip channels. I was looking into the debug and ,even if I have the line accountcode=XXX into the users sections of my sip.conf, I don't see it logged into the cdr. Matteo Brancaleoni mbrancaleoni@espia.it Emmegi System Administrator EspiA - EMMEGI Srl - e*solution provider Uffici: Via Pascoli, 37 20129 Milano - Italy Sede Legale: Corso

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

Hello, I'm working on some dialplan rules to pull multiple users into a conference call. I have some fairly straightforward rules which start up a new MeetMe conference, allow escape with the * key to invite more users, then use a features.conf sequence to bring the new user into the conference with ChannelRedirect. The problem I'm running into is the time in the MeetMe conference

Hi. here is a brand new cdr_mysql . Is based on the previous one, but with lots of changes, so I include here the whole file, not a patch. What I've changed: added new fields, in order to reflect cdr csv : * call start time * call answer time * call end time * call unique id Changed the table structure to reflect cdr.h lengths. Added some (a lot?) sanity checks, to be sure to insert

correct for a transfer. In the traditional Telco World the src (or A Number) field tends to be both the callerid of the customer and an identifier that ties the CDR to the customer for billing purposes. With Asterisk and a lot of other modern day softswitches there's usually a field called accountcode or similar which can be used to tie a CDR to a customer. The src field is then only

My cdrs are missing accountcodes for incoming calls from other asterisk servers.. I've seen a few people mentioning this on the list and the solution seems to be setting up a dialplan for incoming calls from a particular sip peer.. in my opinion this does not scale well at all and I am looking for a solution to correct this problem. example sip peer: [asterisk_gw] type=friend

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

I have defined a SIP friend without username and secret, only IP-based. I have also defined an accountcode for that "friend", as follows:   [mypeer] type=friend host=192.168.0.100 port=5060 context=mycontext canreinvite=no accountcode=mypeer   Unfortunately the accountcode for the calls originating from "mypeer" doesn't show up in the log (either CSV or ODBC). All the other

I have a snom 320 connected to an Asterisk server. I do some weird things using the AccountCode as an identifier. When the snom makes a call, the AccountCode is used successfully in the dialplan as a variable ${ACCOUNTCODE}. When that same call is transferred using the button on the snom, I see a SIP REFER message being received on the * server and the call is transferred - however, this

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

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'm trying to connect an asterisk server via oh323 to a Lucent iMerge. I patched the code due so that Lucent can handle asterisk's ver4 h323 http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration I can now successfully dial in to our company over multiple lines. The issue is when I dial out The first outgoing call connects to an outside user A The second call drops the first

I am experiencing a very wired problem. Some of my cdr are lost. I use logging cdr to csv, mysql and odbc. But some of them are lost. They miss in csv mysql and odbc, so i'm pretty sure it is related to asterisk functioning. I am running asterisk 1.0.7; this is simple configuration file: extensions.conf [general] static=yes writeprotect=no [macro-gw-voipjet] exten =>