similar to: Sipura Blind Transfer - Help

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, 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 List, I hope this setup must be done by our astersik users.. I am using Sipura 3000 to receive PSTN calls and forward those calls to asterisk for voice processing and after that, I am transferring call to extension through FXS port on SPA 3000. Currently, media of call is trombone through asterisk. i.e achieving blind transfers on asterisk with SPA 3000. Is it possible to stop trombone

Yes, After call transfer,I don't want to be media go through Asterisk. Is it possible ? Thanks, Karun. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Dameon D. Welch-Abernathy Sent: Thursday, August 12, 2004 12:07 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Blind Call Transfer using

Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer

Dear list I have managed to write a short program to evaluate data which is inputted from a csv file using the following x = read.csv("wms_results.csv", header=TRUE) All works fine but since I have a number of similar data files which have different names, I would like the program to allow me to input the file name, rather than having to edit the program. >From the documentation I

Hello, I have an r function that creates the following dataframe tresults2. Notice that column 1 does not have a column heading. Tresults2: [,1] estparam 18.00000 nullval 20.00000 . . . ciWidth 2.04622 HalfInterval 1.02311 pertinent code: results<-cbind( estparam, nullval, t, pv_left, pv_right, pv_two_t, estse, df, cc, tbox, llim, ulim, ciWidth,

On Thu, March 5, 2015 09:56, Ruben R?gels wrote: > > Hi again, > > I'm glad to hear that I provided a somehow useful answer. > > Unfortunatelly, I don't know these details. > If you wasn't lucky consulting the snom docs, maybe the snom support > can be helpful with information about the exact implementation > details. > > You also could use "sip

Hi Guys I have an issue where a call is picked up from a queue. The caller asks the person who answered to attended transfer to extension 3082 (for argument's sake.) 3082 picks up the attended transfer and speaks with the outside caller picked up initially from the queue. A few seconds after 3082 has started speaking to the outside caller - 3082's call goes dead in their

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

How well dose theora handle varaiable frame rate? I looking a using a rolling frame rate that move up and down from 1 frame per 60 second (no montion) to 100 frames per second (very fast montion eg: lightning) tom_a_sparks Light travels faster then sound, which is why some people appear bright, until you hear them speak

Hello! I'm trying my hand at lattice representations; I would like to represent a continuous varaiable as function of 2 factors and therefore use the following: bwplot(x ~f1| f2) which works fine except that it plots black points at the value of the median. How can I remove them? Thanks Anne [[alternative HTML version deleted]]

Hi all, I've done some minimal searching on this topic, but haven't come up with anything conclusive. Right now we're using WAV to store voicemail messages, that then (for the most part) get sent to users in email when they have new voicemail. The reason for this is that we have a very mixed environment, with most people in linux, some in OS X and a couple in Windows, and I

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, 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: --

CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime? It appears that RealTime for the extensions.conf file is on a context by context basis, but you have to create each new context in the extensions.conf file then add a "switch => Realtime" line (then reload). I want to be able to add phones without having to edit any files.

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

I just put in a Audiocodes Mediant 1000, which seems to be working well except for one annoyance. I am using Polycom 501's and 601',s and if I do a supervised transfer of a PSTN call where I complete the transfer before the 3rd party has answered, the PSTN party hears dead air until the call is answered or goes to voicemail. I'm not really sure where to start my troubleshooting. Any

I am trying to configure the FXO port on a Sipura-3000 for use with Asterisk. When I connect to the Sipura to dial out on the PSTN line connected to the Sipura's FXO port, it gives me the dialtone of the PSTN line and then I can hear the DTMF for the number I dialled beforehand. It does work but the customer perceives this delayed second DTMF feedback as "unprofessional" and the

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear