similar to: Set(LANGUAGE()=language) - for queue

Before I relate the actual problem, some context. Callcentre environment, a few users testing a new digital dialer... 1. Agents are using Grandstream ATA HT486 and a small analogue dialpad with a headset. 2. SIP connection to Asterisk-1.2b1 3. IAX2 connection to ITSP provider. The call is initially set up in the following way. 1. Agent calls into a meetme conference room and subseqently stays

I have installed several call centers in the netherlands with the eyebeam softphone (from the counterpath guys) It is not free, but very stable, and pretty easy to use. It works great with asterisk (specially the presence option, so agents can see whether somebody is actually ready to take a call). In combination with sennheiser headset CC series, I have had no complaints. We also use a tapi

Hey Guys We are replacing a BM4 with a trixbox (asterisk) virtual numbers as the customer wants to move the callcentre. They are asking for an equiv to the ipview I gather HUD may be or the panel view The problem is that we need to see (a) total calls in the queue (b) calls for specific DID - How can you give 1 DID preference to another DID ie DID 61740410001 = Fred Electrical DID

Hello People, I am a newbie asterisk and happy user, i have configured a x100p card and everything works nice, i can forward incoming connections to a x-lite software client and works out of the box, However when i try to make a connection between two x-lite clients then no audio is transmited, i have followed the instructions on voip-info.org, the tutorials on onlamp and i have read some

Hi ... I am desperate now to solve this problem ... this is my absolute last resort attempt! I have read of the problem on a number of posts/newsgroups/webpages - but there a very little solutions - or causes - none of the solutions have worked for me! Basically my samba is happily running and I can connect to it fine from 9x clients and from 2k client as a workgroup. However..... If I try to

I use asterisk 1.2.5 and h323 that comes with addons 1.2.1. Incoming call comes on h323 trunk. Person A (local SIP phone) receives call and tries to make attendant transfer to person B (local SIP phone). They speak. Then A hangs up. Call form h323 trunk doesn't get to person B. This is what I get on CLI. -- Attempting native bridge of OOH323/xxx.xxx.xxx.xxx-5381 and SIP/307-5663 --

I'm wondering if there's a way it will show on the phone when there's a new message. Here's what I'm trying to do : in my extensions.conf when someone call from a PSTN line on my TDM04B card they have a choice. When someone press 1 for sales then I have 3 phones ringing at the same time. Each phone as already there own mailbox because if someone know there extension

Le mercredi 19 juillet 2006 ? 20:05 +0800, Janus.Wang@QMITW.COM a ?crit : > Hi Jean-Marc, > Please read the attachment file for Marvell 8618 datasheet for your reference, Too much is like not enough. Care to summarize? (speed, arch level, ...) > BTW, Could you mind to specify the voice computing power for Speex codec? > 1. Sample Rate 8K / Bit Rate - 2.15K ~ 24.6Kbps ( NarrowBand)

Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk....? If this call is coming from a queue, do not follow a

Hi list! How to send a call directly to voicemail recording? When I put this exten => 313,n,VoiceMail,u221 Or this exten => 313,n,VoiceMail,b221 In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible? -- Tomislav Parcina tparcina#lama.hr

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

Hi list! I need help with distinctive ring on Cisco 7940 phone. I'm using Asterisk 1.2.5 (I know, I should upgrade) and in dial plan I have: exten => _64X,n,Set(_ALERT_INFO=Chirp2) exten => _64X,n,Dial(SIP/${EXTEN},30,wWtT) On Cisco in Settings => Ring type I have "Chirp1" and "Chirp2". By default phone is ringing sound "Chirp1". For internal calls

I have downloaded samba-2.2.2-sparc-solaris-2.8.pkg from your site and would like to install it to my unix server sun solaris 5.8. I am battling to use the pkgadd to install this package. Can you please help me in this regard as I have failed to find samba/docs/htmldocs/UNIX_INSTALL which is suppose to be instructions on how to install it on unix sun solaris. Thanking you in advance. Regards,

Good evening, I just signed up with Nufone and I am able to receive calls with no problem via my 800 number. Outgoing calls are not going through though. My extensions.conf is as follows: [nufone-out] exten => _91NXXNXXXXXX,1,SetCallerID(mynumber) exten => _91NXXNXXXXXX,2,Dial(IAX2/user:pass@switch-2.nufone.net/${EXTEN:1}) exten => _91NXXNXXXXXX,3,Congestion Whenever I try to

Am 23.01.2015 um 08:47 schrieb Kevin Laurie: > Dear Marc. > Thanks =) > Already have dovecot-solr installed. > Is there a way to see if dovecot-solr is actually working? Yes. http://wiki2.dovecot.org/Plugins/FTS/Solr - look there at "Testing." Use a mailbox for it with at least some megabytes of data. If you don't see then upgrade notices, it it not working yet. The

Hello, I am Janus of QMI?s PLM, We would like to implement the Speex Codec in our embedded system for VoIP Product application, Therefore, I have few question and listed it below, Q1, how much of MIPS we need to reserve the system computing power for Speex Codec needed? Q2, do we need to use the voice DSP for Speex Codec needed? Sorry to bother your job Best Regards Janus (

Hello R-Group, I am trying plotting simple time-series with ggplot2 because the output looks better and I've heard its richer in features. I have the following dataset. > dput(dat) structure(list(Date = structure(c(14970, 14971, 14972, 14973, 14974, 14977, 14978, 14979, 14980, 14981), class = "Date"), Close = c(5998.1, 5996, 6060.35, 6101.85, 6134.5, 6157.6, 6146.35, 6079.8,

Hi Jean-Marc, We already decided to use Marvell's 88F8618 platform, But I have few questions to you and listed it below, Q1, Can we implement the Speex Codec on Marvell 8618 for non-HW DSP of platform? Q2, Based on ARM9 CPU core, do you have any information to let me estimation the voice computing power for Speex codec? Sorry to bother your job again.... Best Regards Janus ( Freedom )

Hi, I rebased rw-devices patch for rsync last weekend. Seems to me it works "incrementally". Attached are * six git commits, or * one complete patch on top of current development rsync branch (master branch commit 60ef39705797c9df7069297eb4ed5feab5e88f29). bobek:/data/soft/rsync/rsync# dd if=/dev/urandom of=/dev/mapper/vg-pokus1 dd: writing to `/dev/mapper/vg-pokus1': No

In samba source code, the file "source\include\msdfs.h" defines a struct named referral.In this struct, there is a field named ttl which is used to decide how long should client cache referral, and this field uses constant REFERRAL_TTL(#define REFERRAL_TTL 600) as default.My problem is when i set REFERRAL_TTL to be 10, it doesn't work.