similar to: Understand SIP REFER

Hi! I have the following problem with Asterisk 1.4.23: ATA w/ T.38 Asterisk ATA w/o T.38 --------INVITE--------> --------INVITE--------> <-------200OK---------- <-------200OK---------- --------ACK-----------> --------ACK-----------> --------INVITE

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

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 guys, I am wondering (and have been unable to find out thus far) whether Asterisk sets some special channel variables or something when a call is transfered with the REFER method. Basically, I'm trying to figure out if it is possible to somehow get a transferred call back to the transferrer (as it is done with the built-in atxfer) after X seconds (or an unsuccessful attempt). Using a

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

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

OK, it's been a while since I drank from the pool of wisdom hear on the list. After cracking my head against the wall for a few days trying to figure this out, I have decided to swallow my pride and take the drink. So, on to my question: I have some agents/operators setup in sip.conf which point to a context where I have just about disabled outbound calls (only specific numbers can be

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

Hello! Most are probably bored seeing another letter about this, but I've put in a fair amount work on a spec for rewriting the CDR system in Asterisk, and I have some questions: First, please look at what I've written so far: svn co http://svn.digium.com/svn/asterisk/team/murf/RFCs and look at the file "CDRfix2.rfc.txt" in the RFCs dir. The spec SIGNIFICANTLY alters the way

Is there a general recipe to avoid fraudulent calls under the following conditions? A receptionist transfers calls as a callee (customers are calling) and as a caller (boss asks to call and then transfer to him), i.e. the Dial cmd for the internal context contains "Tt". Then an outside call would operate as a Local channel in an internal context after the first transfer. If the

Hi list! When I'm doing transfer, to what context/priority does that call goes? Can it be changed? Is it the same for blind_tr/att_tr/and for transfer that appears when phone replies with - 302 "Moved Temporarily"? The thing is that I'm trying to transfer incoming call from E1 interface back to E1 interface. Transfers will occur when user is going out and sets up all call

Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all -- /*************/ nik600 https://sourceforge.net/projects/ccmanager

Hello, while most of the physical phones have keys to handle attended and blind transfer, most soft phones have no support for it. Asterisk offers a "featuremap" to assign a key to blindxfer and atxfer and they work fine if the call is still in the same starting context, but if the call has moved in another context, then the new call will be started from such context with unpredictable

Hi, Does anybody know if there is a way for a call goes back to transferer if unanswered ? Thanks Luis A P Barbosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070912/1e356013/attachment.htm

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 all, I ve setup a queue with 2+ agents for managing our inbound calls from customer. Using Asterisk 1.2.18 in a CentOS box. Agents login using AgentCallbackLogin application and I use a BASH AGI to accomplish this as there are some validations done with MySQL DB. Im aware that transfer could be done with option 't' in the queue() application and I was able to successfully transfer

Hi, You can rebuild the RPM on a CentOS 8 box from https://dl.fedoraproject.org/pub/epel/7/SRPMS/Packages/s/squirrelmail-1.4.23-1.el7.20190710.src.rpm or https://dl.fedoraproject.org/pub/fedora/linux/releases/31/Everything/source/tree/Packages/s/squirrelmail-1.4.23-2.fc31.20190710.src.rpm and install the resulting RPM. For the source https://sourceforge.net/p/squirrelmail/code/HEAD/tree/

Hi All, I need to stop the transfer feature on particular sip user. I am using linksys phone and it has set the forwarding enable to another user. I have three users 2101, 2102, 2103. 2102 is registered in linksys phone with forwarding enable to 2103. But is there any procedure in asterisk that we can not allow 2102 not to forward on 2103. and also i want to prevent the SIP/2.0 302 Moved

Hi All, I need to setup asterisk to receive fax. I'm try Spandsp (opensource) and Attrafax (commercial) both on asterisk 1.4.23) but the results are disappointing. with spandsp many times the fax arrives cut. with Attrafax i have some problem. Anyone have any idea or solution (Opensource or commercial) to suggest me ? Best Regards

If I understand correctly, the desire is to change the current behavior in a kind of subtle way. Is there some way you could instrument trunk's FileCheck to die if the old behavior is encountered, and use that to definitively find all tests which rely on the current behavior and migrate them (possibly to an interim solution) in preparation for the change in semantics? How much is the old