similar to: Issue with Asterisk not hanging up second leg when first leg hangs up

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

The Asterisk Development Team has announced the release of Asterisk 1.8.2.3. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.2.3 resolves the following issue: * Reimplemented fax session reservation to reverse the ABI breakage introduced in r297486. (Reported by Jeremy Kister on the asterisk-users

I'm confused about something. It's the way Asterisk handles the A leg (ie the first party dialed) on an originate command via the Manager Interface. Lets say our originate commands looks like this: ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/5551212 at provider Variable: destination=SIP/8675309 at provider Callerid: 5551212 Context: default ActionID: 849120

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, The following patch provide LDAP public key authentication instead of only ~/.ssh/authorized_keys. Allow you to centralize/control easily users authentication on a server farm. Patch: http://ldappubkey.gcu-squad.org/ldappubkey-ossh3.5-2.patch more informations on http://ldappubkey.gcu-squad.org/ Best Regards, - -- Eric. -----BEGIN PGP

Hi, When implementing click2dial feature, I can trigger an Aastra phone to auto-answer using statement like : SIPAddHeader(Alert-Info: info=alert-autoanswer); This is very convenient when trying to reach a distant party (ie through PSTN) The trouble is when 2 Aastra are calling each other over the LAN, this single statement is memorized somehow and both phones (caller and callee) auto-answer.

Hi We have the following test .call file and test dialplan: I can't set a custom CDR var to a value on one channel leg, and another value on the connected channel leg? Is there a way I can woraround this issue? ## test call file Channel: Local/queue at TiagoGeada CallerID: teste-geada:0:210332450: MaxRetries: 0 RetryTime: 1 WaitTime: 8640 Account: teste-geada Context: TiagoGeada

Hello friends, I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can access the caller Call ID (fbasename field in voipmonitor cdr) looking at the SIPCALLID variable in asterisk, but how can I access from within asterisk the Call ID of the second leg of the call (the one originating from asterisk to the destination peer)? is there a variable holding this value? Thank you

Hello all, I would like to set a different caller-id for the second leg of a call when doing an originate. For example: Action: Originate Channel: sip/1234 Context: mycontext Exten: 1 Priority: 1 Callerid: "123 <123>" Async: true This sets the caller-id correctly when dialing sip/1234, but I would like to set the caller-id for the second leg of the call (the one that goes to 1 at

Hello All, New version of the openssh ldap public key patch is available at : http://ldappubkey.gcu-squad.org/ Changes : Full RSA1/RSA/DSA options+key supports LDAP based group management Multiple keys per user Bugfixes + Partial patch rewrite Hope it helps, Best Regards, Eric. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:

Hi Group, I have following case scenario. Through call file, Asterisk makes a call to SIP extension. When Extension answers the call, Asterisk reads customer numbers (set in callfile) and calls them one by one untill one of the customers answeres the call. Here customer and SIP extension gets patched and talk to each other. Now if outgoing call is answered by Answering machine,I don't want

Hi, I have made new small patch. He use X509 certificate to authenticate users. This patch use some features which are coded by Eric Auge (see ldap patch http://ldappubkey.gcu-squad.org/). You could find the patch on http://traceroute.free.fr/articles.php?id=24 regards, Fred.

Hi all, i am wondering if i can make two leg cdr in mysql cdr table. 1st Leg : Registrar the ATA which registered to the asterisk and it normally logging in cdr table. 2nd Leg : The CDR of carrier for the example if i send call like exten => _x.,1,Dial(SIP/${EXTEN}@AT&TIP) I this cause i can get the accrue duration of call because currently we are facing some call missing not coming

Hi list, We, an OSS advocacy group, setup a Xen 3.1 machine composed of : . a 64 bits dom0 running Debian stable amd64 . 2 hvm domUs running OpenBSD amd64 . 2 hvm domUs running NetBSD i386 This machine is to be hosted and reachable from the Internet, but it will only have one public IP. Naturally, our first tought was to port-forward using iptables / netfilter. We didn''t really

Hello My ISP provides an FXS port to plug a handset, which can be used to make free calls to (GSM) cellphones, similar to the Billion ADSL modems: http://au.billion.com/product/voip.php My plan is to install an SIP client on a smartphone, so that when I'm travelling, I can connect to a good wifi hotspot, register with an Asterisk server at home which has an FXO card, tell Asterisk the

Hi, I'm using Asterisk to bridge the incoming call to another destination using the Dial command. However, when an anonymous call comes in then privacy information is not passed into the B Leg. For instance, the Privacy header and P-Asserted-Identity aren't copied to the B Leg. Is there an option to give to the Dial command, or another variable to set, to make Asterisk copy such

Is there a way to specify the outbound leg Call-ID? -- Eric Chamberlain

Hello there, I'm using a Debian box with Asterisk 1.8.13.1 as a DID-PSTN gateway, so I can receive calls in a DID number and redirect it to my mobile line. It has been working flawlessly for a few months, but I have noticed that some calls were not cut after losing one leg (the one with the DID server), and kept the PSTN leg active until I restarted the server (with the unexpected cost

I am sending this problem for 2nd time. Please help. Thanks _____ From: Omar McKenzie [mailto:omckenzie@trenetinc.com] Sent: Thursday, September 08, 2005 9:57 AM To: 'asterisk-users@lists.digium.com' Subject: Problem: Got SIP response 481 "Call Leg/Transaction Does Not Exist" I am not able to get softphone registered (active) with * . new installation , new user

For a simple call file like Channel: Zap/g1/XXXXXXX RetryTime: 60 WaitTime: 30 Context: from-file Extension: s Priority: 1 I noticed that s@from-file started to execute regardless of the state of the outgoing call. Is this supposed to be? So far I can only set a Wait() in the local leg and hope the remote party picks up soon enough. I thought call file extension will start execution only

I am not able to get softphone registered (active) with * . new installation , new user Able to get server started , and phone appears to register . gets the SIP reponse 481 message Register SIP '4009' at 192.168.200.10 port 2199 expires 120 Unregistered SIP '4009' Register SIP '4009' at 192.168.200.10 port 9428 expires 120 Saved useragent