similar to: IAX/SIP Inter Asterisk Transfer

I've got the ztdummy kernel module loaded and seem to have all the desired prerequisites in place, but Asterisk never seems to compile with MeetMe() application support enabled, nor does there appear to be a module I am failing to load that would contain this application. Is there something really obvious I am missing? Thanks, -- Alex Balashov Evariste Systems Web :

Hi, I want to check the status of a blind transfer (only sip endpoint) between various phones. Transfer is working perfectly, using ## from features.conf or using transfer key from phone, here SNOM320. My problem is that if party to transfer to is busy, the transfer fail and the call is ended. What I want to do is to return the call to the party who originate the transfer. I checked

Hi, Can anyone suggest a way of doing the following in Asterisk 1.6.2 - I do not think it can be done trivially using TRANSFER_CONTEXT. What I want is for the TRANSFER_CONTEXT for all technologies to be the same as the initial context defined in the configuration of the device initiating the transfer. This is not as simple as it seems (unless I am missing something). For example: A call arrives

Hi all, when doing a blind transfer using the keys defined in features.conf, we hear a confirmation of the attempt to blindly transfer, followed by an invalid extension message. The console says this: [Jun 4 22:30:31] VERBOSE[11301] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/570-00000006 [Jun 4 22:30:31] VERBOSE[11301] file.c: --

Hello, I'm trying to find more information about this Remote Attended Transfers, as is explained in https://wiki.asterisk.org/wiki/display/AST/res_pjsip+Remote+Attended+Transfers for Asterisk 12 using pjsip stack Was Remote Attended Transfer implemented in previous versions of Asterisk (versions without PJSIP, Asterisk 11 and previous)? Where can I find configuration examples to do it work

Hi, We are running asterisk 1.2.16 and need to connect two channels which are already established. We are currently using app_meetme to achieve that, but we are sometimes unhappy, as app_meetme provides functionality that produces load that we do not need in our two party conferences. I figured out that there is an alternative called app_changrab.

Hi, Anyone using inter Asterisk trunking IAX /IAX2 ? Thanks, Geo ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

I originally wanted to answer with something ... tzarit and kevit Readed probably before you invent rapid biz. I am asking to share any info /experience not your high spirit. Thanks for less trivial answer, G >On Wed, Oct 05, 2005 at 12:44:27PM -0700, Thameem Ansari wrote: >> I am using the inter asterisk trunking and the article in >> voip-info.org<http://voip-info.org>will

Hi all, lets say I've registered at several Sip-Providers. Provider A offers best rates but is often too busy to get a line. Sip Provider B is stable (but more expensive). The asterisk box has a high call volume therefore problems at provider A will get obvious after a few calls stalled. In this case astersik shall switch temporarily to provider B but shall test periodically for selected

Hi, I am in a weird situation where a variable seemed ignored, but not always. That variable is __TRANSFER_CONTEXT. Basically, I have a phone registered with asterisk. It's context is "internal". Outgoing calls go through that context (all good). When I get an incoming call which I want transferred, I don't want it to go through the context "internal" but

Hi, while trying to sync the mailboxes of several users who use and share their mailboxes dsync prints this message: dsync-local(<user-who-uses-shared-mailbox>): Warning: Subscriptions file /home/<user-who-uses-shared-mailbox>/Maildir/subscriptions: Removing invalid entry: shared/<sharing-user>/<shared-folder> The problem is: Every user has to subscribe the shared folder

Hello, We are in the process of splitting our phone system into two separate logical systems for our two departments. One of the goals of this switch is to restrict members of one department from transferring calls to the other, but not restrict them from calling that department themselves. So what I need to know is how to detect whether a call from a member of that department is a transfer or

Hello list, I have a problem with called ZAP channels making an attended-transfer or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk is wrong. At the moment there is a bristuffed Asterisk 1.2.18 running with bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit: [default] exten => 0123456789,1,Macro(dialpstn,${EXTEN}) [macro-dialpstn] exten =>

Hi, I recently upgraded to openSuse 12.1 which comes with dovecot 2.0.14. Because of mail-client-problems I am running one dovecot which requires authentication via a client-certificate and another one which can be used without a certificate. (Configurations can be found below.) Since the upgrade our shared mailbox is no longer visible. I tried to repair this by setting the ACLs once again (using

Hi all, There's been a few posts looking for telemarketer torture scripts so I figured that I would write one using a SQLite db. Handling an incoming call that is flagged from the database is pretty simple. My problem is that I would like the callee on an established channel to be able to redirect the caller to a specific context where my AGI is called and handles the call by first

<div><FONT size=2>Hello,</FONT></div> <div><FONT size=2>two questions: </FONT></div> <div><FONT size=2></FONT>&nbsp;</div> <div><STRONG><FONT size=2>1: How can I open/enable network connection to B?</FONT></STRONG></div> <div><FONT

I am trying to setup a couple of virtual pbx's off of my one may asterisk box. So far I have been able to segment most everything via the Dial plan. My only question/problem has to do with the # Transfer function. I had set up # Transfers prior to segmenting the dial plan, and I cannot remember how I was able to specify which context to use when the user presses #. I haven't been able

Hi all, there is an application called changrab with quite interesting capabilities: http://www.freeswitch.org/asterisk_stuff/app_changrab.c I think it is available in the 1.4 version by default!? This application can connect to channels which are already UP. The only possibility AFAIK to connect channels after they are UP are the well known conferencing applications meetme and conference. If

I know this isn't an asterisk thing, but since the recommendation to get one came from here I figure lots of people out there have one. I read the docs, and it says that in order to do a blind transfer I should hit "flash", then dial "*__" then the number. Now, how on a normal phone do I dial "asterisk underscore underscore"? Can someone tell me how doing a

Hi, I am trying to understand why some of my call transfers fail. My scenario is as follows: Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2 Step1: PBX1 extension 101 calls PBX2 extension 102 Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension 103 Step3: PBX1 extension 103 answers the call and transfers it to PBX2 extension 104 Step3 fails and extension 103