similar to: Connection problem w/ Attended Transfer

Hi Guys I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest of the ports in their own groups so that I can have different signaling on each? [channels] language=en switchtype=euroisdn pridialplan=unknown resetinterval=600 echocancel=yes echotraining=yes

We have a customer on a wireless connection that has very bad jitter. They can hear people fine, but people have a very hard time hearing them. They are connected via a SPA-2102. It is a SIP client going to a SIP trunk. Something like this in sip.conf [general] would be in effect for all SIP clients: jbenable = yes jbmaxsize = 150 jbresyncthreshold = 1000 jbimpl = fixed jblog = yes I only want

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

4 Port PRI sangoma a104 From: jg [mailto:webaccounts173 at jgoettgens.de] Sent: Wednesday, March 18, 2015 2:09 PM To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 4 Port PRI I have a 4 port PRI card that I need to setup each port in their own group. In chan_dahdi.conf I have the following which works for one port How do I add the rest

Hi all again, I've got a problem with sporadic one way audio calls, which means sometimes I can't hear the calling party (call is established, but audio is missing). Today I received ~90 calls, one of them got this problem. I've got two networks involved, without NAT: - 192.168.1.X, in there one nic of my server and all the phones - a private net to my provider, in there a nic of my

hello every one, i have an asterisk system and want to act as gateway and send calls to cisco gatekeeper. this is my h323.conf file: [general] port=1720 binaddr=192.168.0.YY context=from-trunk faststart=yes h245tunneling=yes gatekeeper=192.168.0.XX //cisco address progress_setup=8 progress_alert=8 dtmfmode=rfc2833 jbenable=yes jbforce=no jbmaxsize=200 jbresyncthreshold=1000 jbimpl=fixed jblog=no

Hello guys, I am having some problems with calls comming from the PSTN lines, when somebody calls people can't hear me, but I can hear them, every day I have to do a /etc/init.d/asterisk stop && /etc/init.d/dahdi restart to have calls with sound again, wich cli dubug commands can I use to see what is going on, here I have my chan_dahdi.conf and sip.conf, I am using 1.6 Thanks a lot!

Hi all, I'm fighting with a really strange problem that is really busting me. I have an asterisk 1.4.22 ( from a trixbox 2.6.2 ) and mISDN 1.1.7 3 extension on hardphone and 3 extension in softphone ( zoiper ) What happens is that sometimes the people on the other side of communication hear my voice as metallic and chopped. This happen either on incoming call than on outgoing call. If I

> I have a 4 port PRI card that I need to setup each port in their own group. > > In chan_dahdi.conf I have the following which works for one port > > How do I add the rest of the ports in their own groups so that I can have different signaling > on each? > > [channels] > > language=en > > switchtype=euroisdn > > pridialplan=unknown > >

Hi, do you have NAT between Asterisk and agent phones? S pozdravem Tomáš Holý Hi Tomas Thanks for replying. Yes, the phones are in one location in a LAN and are then NATed to enable them to contact the Asterisk which is hosted in the cloud. A typical sip.conf phone configuration on the remote server for the site is [general] session-timers=refuse disallow=all allow=g729:20 allow=ulaw

I''m having trouble getting anything in the params hash when I do: <%= form_tag(:action => ''edit'') %> <table border="0" id="detailtable"> <tr> <td>* Email:</td><td><%= text_field_tag(''member[email]'') %></td> </tr> <tr> <td colspan="2" /> </tr>

Hi, I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also. But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized

The Wine maintenance release 1.2.3 is now available. What's new in this release (see below for details): - Translation updates. - Various bug fixes. The source is available from the following locations: http://ibiblio.org/pub/linux/system/emulators/wine/wine-1.2.3.tar.bz2 http://prdownloads.sourceforge.net/wine/wine-1.2.3.tar.bz2 Binary packages for various distributions will be

Hello! I use asterisk with TE420 as PRI switch for two channels : ;panasonic uplink group=3 context=panasuplink ; relaxdtmf=yes ; immediate=yes rxgain=0.0 txgain=0.0 mohsuggest=default jbenable = no ; jbenable = yes ; jbmaxsize = 200 ; display_send=name_initial display_send=name

Hello, any one advise a good, strong and free softphone that can work with SIP or/and IAX lines and supports attended transfer ? Thanks for help. Mit freundlichen Gr??en / best regards Andr? Herrlich IT-Operator / Developer ____________________________ LetMeRepair LMR Service and Consulting GmbH Fichtestr. 1A 02625 Bautzen Tel.: + 49 - (0)3591 - 2722 - 1451 Fax: + 49 - (0)3591 - 2722 -

Hi, Since we're moving from a legacy PABX that has been serving one of our customers for more than 15 years, we'd like this process to require no "human habits" change among the users. Software: Asterisk 1.4.22 Hardware: Polycom phones (mainly 430/601) Here are the "problems": We did configure call groups, pickup groups, ... - When someone picks up a call from

Hi All, I see some patches about adding atxfer beep sound in the sip channel, but I'm not clear on when this was implemented in what version? I don't see the added function in chan_sip in 1.2.24 or 1.4.21 or 1.6.0.28? Where is this code implemented, what stable release? Thanks. JR -- JR Richardson Engineering for the Masses

Hello, I'd like to implement something similar to an attended transfer, but with a little more control (I'd like to be able to use MixMonitor and StopMixMonitor to control the call recording, set the account code, etc. I'm on Asterisk 1.4.26. All of the ways I have seen to do this are complicated plans using MeetMe and applicationmap features, and playing with those over the

Hi, Asterisk: 1.8.4.2 I've just managed to configure attended transfers using Asterisk and Grandstream GXP-2000 phones. The only way I've got it to work is by using one of the out-of-band DTMF modes on the phone (either RFC or SIP-info). I think I can understand why - as Asterisk wouldn't be "seeing" the DTMF tones during the call if they are inband (or am I wrong)? I

I am looking for a way to have an agent execute an attended transfer using the asterisk manager interface (AMI). I have been trying to use the dual Redirect from svn trunk. The problem with this function is that the "ExtraChannel" does not get redirected properly afaict. Now, I am looking for other solutions for the list, I will probably try playing DTMFs on the agent channel to