similar to: Can I start recording channel in the middle of conversation ?

> -----Original Message----- > From: Robert Rozman [mailto:rozman@fri.uni-lj.si] > Sent: Monday, January 17, 2005 7:45 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Can I start recording channel in > the middle ofconversation ? > > > Hi, > > I'd kindly ask for simple example if this is possible ? > > Is

Hi, I have following problem. Asterisk is connected to ISDN router on BRI interface. ISDN PBX is connected to another channel of BRI interface. Now I'd like to route all incoming calls first to Asterisk and then if caller wants to talk to extension on ISDN PBX then I'd like to route call to another capi channel but free the current one. Is this possible at all or do I need to take 2 capi

Check your FW-1 tracker and see if any sip packets are dropped during call initiation. I had this problem and it went away when I upgraded the BT's firmware to the latest (16). Beware, though, that people on the list claim that this firmware breaks functionality of the message button and autoanswer. I haven't checked this yet, cause I can't afford to go back a version. I prefer a

Hi, I'd like to setup oneway connection - so asteriskB can place calls on asteriskA and be safely authenticated with rsa keys. I just don't get any response on asteriskA. I've generated pair of keys: name.key, name.pub and put them on both servers - is it right to only have name.key on asteriskA and name.pub on asteriskB ? I get everybody is busy ... on asteriskB, and none

Hi, I'd like to trigger call recording during call. Do I have any keys that can be pressed during call ? I've tried this, but doesn't start anything ( I guess that "a" is active only during voicemail ?): exten => a,1,DBget(temp=Record/${TIMESTAMP}_${UNIQUEID}_${CALLERID}) ; Already recording ? if not goto 102 exten =>

Hi, I'm trying to implement dynamic routing of incoming calls to local extension if previous outgoing call was unanswered. But after I do Goto to s-NOANSWER, variable ${EXTEN} changes to 's-NOANSWER'. I guess this is normal, but I don't understand why ? How to workaround on this one ? Thanks in advance, regards, Rob. [outbound-capi-ISDN] exten => _0.,1,NoOp(Calling ISDN

Hi, I'd like to use DISA properly for my case - I'd like to handle it right, if user when in DISA doesn't dial any number - how does Asterisk return from DISA cmd ? I'd like to dial some default number if user doesn't dial anything or give him some message - but I don't know what gets executed after DISA if nothing is dialed .... I'm reading this on wiki, but

Hi, I noticed that agents logins (agentcallbacklogin) are reset if Asterisk is restarted. Can this be avoided in some way ? Regards, Rob.

Hi, I've found approximate same pricing for both. Sipura 2100 seems to have more features... What are differences between those two ? What about their reliability (specially regarding fact, that they deal with analog phones) ? Thanks in advance, regards, Rob.

Hi, what would be general choice between those two options? They are related to two different things: security, performance. But what would I loose if I run as non-root without realtime priority ? Thanks in advance, regards, Rob.

Hi, at the university department we have quite old Panasonic 2+6 PBX (1BRI + 6 POTS for outputs) and 25 local analog extensions. We would like to add Asterisk with 1 fresh BRI line and possibly integrate with existing equipment (we would like to crossover between both pbxses). What would be most efficient way to do this ? Thanks in advance, regards, Rob.

Hi, thank you for last info. we've tried to use te110p but failed. We're quite surprised that cable wasn't included with the card as any documentation, at least on HW setup and installation, yet cable pinout for connection to PRI interfaces.... 1. We have followed instructions on your site and from Beronet guide, but card just keeps blinking and nothing happens (also no useful

As far as I can remember I only opened sip and tftp ports for the phone. For some reason (didn't look into it too much) the call stays with sip and doesn't use RTP. The problem you describe (the call doesn't even ring on the other side) is something I had and was solved by upgrading the firmware. Checkpoint's tracker explicitly said what connection attempts were blocked and why.

Hello. I notice that when a call that is recorded with MixMonitor is transfered to another co-worker, the recording ends. exten => 409,n,Macro(SDstartrecording,external,${DID}) the incoming call then goes to a queue... [macro-startrecording] ; ARG1 = incoming DID or CALLERID(name) ; ARG2 = outgoing dialnumber ... exten => s,n,MixMonitor(/var/ftp/${NR}/${recordfile},b,chown -R

Hi, I'm reading that spandsp works only with zaptel channels. What are my options if I want to receive faxes through ISDN Fritz card with Asterisk and possibly forward it as emails ? Regards, Rob.

Folks, I'd like to change the value of ${CALLERIDNAME} for incoming PSTN calls from certain numbers, but haven't found a way that works. The goal is to provide more informative names on my phones' caller ID displays--e.g., I would prefer to display "ROB CELL" instead of "CELLULAR CALL" when I call home from my cell phone. This is what I tried in the context

How do I monitor the whole conversation on a Zap channel without answering it - the channel is hanging up, I think it's because it's not answered.

On 5 December 2014 at 06:49, Sean Silva <chisophugis at gmail.com> wrote: > > > On Wed, Dec 3, 2014 at 4:23 AM, Robert Lougher <rob.lougher at gmail.com> > wrote: >> >> Hi, >> >> In feedback from game studios a common issue is the replacement of >> loops with calls to memcpy/memset. These loops are often >> hand-optimised, and

Hi, PR17975 was caused by r191059 which was reverted on the 3.4 branch in r196521. However, the problem still occurs with trunk (confirmed as of r206186). >From a thread on cfe-commits I see that Kai Nacke (the author of r191059) was working on a patch to fix PR17975, but the conversation ends: http://lists.cs.uiuc.edu/pipermail/llvm-commits/Week-of-Mon-20131202/197968.html So my question

For our helpdesk application, we need record full conversation between any caller and one or two helpdek numbers (while the conversation is running). After conversation is ended (hangup ..), the recorded file (WAW) is putted into database. Using AGI, record and put to database is OK, but only as exclusive task. But I need record WAW file "in background" the standard conversation.