similar to: Transfer (SIP REFER) - AccountCode not available?

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

Hello. I am using a snom 320 running 5.3.6 with Asterisk 1.2.4. In the sip.conf entry, I have mailbox=1234@default and vmexten=*98. The light on the snom 320 turns on when I have voicemail and the retrieve button dials the correct extensions. However, the light turns off immediately after making the call to voicemail, even if I do not check the voicemail. Any idea on how to get this to behave

I have a whole bunch of remote devices connected to my Asterisk box, including SPA-3000s, PAP2-NAs and Cisco 7960s. The PAP2-NAs I have only rolled out recently and I am having a problem that is intermittent and inconsistent. It happens to some users but not other users on the same ISP. It happens to users in 2 different countries where the Internet setup (NAT issues) are completely different. It

> I would like to have all SIP phones to work on prepaid basis > and without need to dial any access number, instead I would > like to use the phone as normal dialing only the destination > number, for example 00464090510. I use the AccountCode for authentication. This is how, for example: exten => _00XX.,1,DeadAGI(astcc.agi,${ACCOUNTCODE},011${EXTEN:2}) > Once the call is

Hello. I was just brainstorming for a future project and was hoping to get some creative ideas from the list. If I have multiple * servers at multiple locations all connected together with a nicely partitioned dialplan (2XX for office 1, 3XX for office 2, etc.) it's pretty straightforward to link them all using IAX and allow intra-office transfers. Further, servers at each location are

Hello. I have set up ASTCC properly, calling it like this: DeadAGI(${ACCOUNTCODE},${EXTEN}) It seems to be working correctly, but I have two questions: - Although the cards' credit seems to be maintained correctly, I cannot see the call details in astcc-admin. When I try to view information on the card, it's just blank. Any ideas? - When does the 2nd, 3rd and 4th trunk get used? I have

Hello. I have my * server set up and working perfectly. I wanted to allows calls to sip:nabeel@sip.myserver.net. In sip.conf, I have: [general] context=default Also, in extensions.conf, I have: [default] exten => myname,1,Goto(internal,nabeel,1) However, when I make a call using a "Direct Dial IP" account in X-Lite, I get the following error in *: Failed to authenticate

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 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

Hello. I have an * server set up on a public IP. I have SIP clients at three different locations, all behind NATs. I have all the SIP users set up this way: [user1] type=friend username=user1 secret=password1 callerid="User 1"<101> host=dynamic qualify=yes context=outgoing All three SIP clients are configured to use STUN (using stun.fwdnet.net:3478). Furthermore, I have

I'm putting together an Asterisk server for a local charity to use as an announcement system. I've been thinking about how to write the dialplan to allow different options for different groups' announcements, as well as mailboxes for the various groups and the charity's administrators. Of course, this would also need to include an option for the heads of the different groups to

Hi, Does Asterisk have a limit to how many NAT'ed SIP clients it supports behind a single IP? I have the weirdest problem ever. I have three SIP endpoints. SNOM phones, if it matters. Their extensions are 200, 201 and 202. Apart from the username/password, the sip entries in sip.conf all have identical configuration. They're all NAT'ed behind the same IP. 200 and 202 registers

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

Is there any SIP or IAX phones that can be configure in french instead of english. I tested Cisco 7960 phones but I can't change the language it's only available in english with the SIP firmware. I have a customer that's located in France and he wants french phones if possible. So I'm wondering if there's any one out there that found a phone that can be change to

Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user

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

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

This is somewhat related to my other query on the list regarding NAT traversal. I have heard many times that IAX is "NAT-transperant". I am unsure how it accomplishes this. I do know that SIP works like this: your SIP device send a request to the SIP server (usually on port 5060) with whatever command. The SIP server respends to your device's "apparent" IP and port (this

> Many thanks Julian. Are you looking for the pinout for a single plug 2.5mm (cellphone) headset or a dual plug 3.5mm (computer) headset? -- Nabeel Jafferali Tel: +1 (416) 628-9342 Toronto +1 (646) 225-7426 New York FWD: 46990 Email/MSN: nabeel<at>jafferali.net

Hello. I am having an issue where sometimes the cheapest provider for certain international destinations is not always reliable in completing calls. However, there is not problem once the call is made (i.e. no lag or echo or anything). The way I have it set up right now (for example) for Dar es Salaam, Tanzania is: exten => _925522XX.,1,Dial(IAX2/livevoip/011${EXTEN:1}) exten =>