similar to: Delay before dialing extension on Zap channel

Hi, I've got a problem with my asterisk set up which has been going on for a while (months). I'm currently running 1.2.7.1 on a gentoo box with the topology below: +-----+ PSTN ---------+ * +------------- Service Provider (wctdm400p) +-+-+-+ IAX | | | | FXS --+ +-- SIP (cisco 7940)

I've got an old Dell Optiplex (Pentium-II, 1998 Vintage) which is successfully running an X100P card. I'm hoping to upgrade to a TDM400. Has anybody tried running these cards in old Optiplex machines? I'm not particularly worried about horsepower - more about the motherboard having a PCI bus that's able to power up the card... -Ronan

2012/1/10, Olivier <oza_4h07 at yahoo.fr>: > Hi, > > 1. This patch didn't correct the issue but I'm far from certain that I > correctly applied the patch. I was right to suspect I was wrong : now, after correctly applying the DAHLIN-275 patch, it's working OK (with the EchoCan module plugged-in). Thanks for your lighting fast correction !! > 2. I took the

Hi, I have configured a certified asterisk 13 server with chan_mobile and res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone client. Now the problem is, using the hardphone I'm able to call the softphone and hear everything properly. But when I call from the hardphone to some number that has to be dialed via chan_mobile, I'm not able to hear what the other side says (I

>Or, you could use a Corinex Phone Line Bridge which runs 128Mbits up to 2000 >feet. They also have a co-ax version which is 200mbits and goes 4000 feet... >About $300 for both ends. too bad they don't say what the bandwidth is at max distance - anyone know?

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, December 01, 2004 7:07 AM Subject: Asterisk-Users Digest, Vol 5, Issue 6 > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Good day, I've got around to installing an X100P card in my computer to try out asterisk. I noticed (and people who were testing with me also noticed) that when dialing from my SIP soft phone to the PSTN, the ringer tone changes after 2-3 seconds, precisely when the Zap channel takes over the call. Is it possible to eliminate the first ringing? Is there a reason to this

Hello, I'm trying to install samba 4 as a DC following this tutorial https://wiki.samba.org/index.php/Samba_AD_DC_HOWTO. Ihave reached the step of testing the connectivity to the DC with smbclient [root at DC-TEST ~]# /usr/local/samba/bin/smbclient -L localhost -U% session setup failed: NT_STATUS_CONNECTION_REFUSED Samba 4 has started successfully [root at DC-TEST ~]# netstat -lntp

king all in vorbisfile make[2]: Entering directory `/home/ronan/libvorbis-1.0.1/doc/vorbisfile' make[2]: Nothing to be done for `all'. make[2]: Leaving directory `/home/ronan/libvorbis-1.0.1/doc/vorbisfile' Making all in vorbisenc make[2]: Entering directory `/home/ronan/libvorbis-1.0.1/doc/vorbisenc' make[2]: Nothing to be done for `all'. make[2]: Leaving directory

Hi, I've just upgraded by asterisk box from 1.0.7 to 1.0.8 / 1.0.9. I'm running Gentoo, and in the UK, on a BT PSTN line. The box has been running more or less fine for several months. Since upgrading asterisk has been failing to hangup inbound / outbound calls. I've kept my original config files. The sequence of events is roughly: - Place call from a cisco 7940 through the

Hi, i upgrade on a new server samba3 to samba4 with a LDAP Backend. I have followed this HowTO ?http://wiki.samba.org/index.php/Samba4/samba-tool/domain/classicupgrade/HOWTO until de classicupgrade step Here is the errors I get ?/usr/local/samba/bin/samba-tool domain classicupgrade --dbdir=/root/samba3/tdbfiles --use-xattrs=yes? --realm=bceao.int /root/samba3/tdbfiles/smb.conf Reading

Hi! I have a very strange question. I'm using trixbox with Asterisk 1.2.23-BRIstuffed-0.3.0-PRE-1y-j. I configured and installed the HFC ISDN card with a script, as here: http://www.trixbox.org/forums/trixbox-forums/help/how-install-hfc-card-trixbox Now i have 6 SIP hardphone, and softphone, using 1 SIP trunk to call out the world, and 1 ZAP ISDN trunk to receive calls from the world. The

I'll have MGCP hardphone that needs to dial pre-defined number as soon as it goes off-hook. So far I'm lost as to how (if at all) this can be implemented in Asterisk. Any pointers? TIA, Peter

Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the

Guys. I would like to hear tips and tricks on extention config best practices, for example, naming, etc. and most of all, how to deal with extention that have a full time hardphone configured and assigned and then a softphone connecting to the same extention, for example, one employee has its hardphone on the office but sometimes when he travel, he uses his softphone to work with, what happens

I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented

Fellow list members, I have run into an issue where I encounter a delay at the beginning of a phone conversation when I make outgoing calls through Asterisk with an Aastra 9133i hardphone. This is most noticable when I call a voicemail system with the Aasta and then with a land line or other VoIP phone. The first word or two of the voicemail message is generally cut off. According to

I'm looking for an enhanced speed dial "dashboard" as DSS (Manager integration) for Operator console integrated in a voip phone (softphone or hardphone, opensource or commercial) to diplay the status of phones (sip, zap, iax...) connected to asterisk. I see in snom site the snom 220 with keypad 220. Can it display the status of internal and external lines (free, busy..) and

I am have same issue with PRI and overlap dialling is not enabled. Stuart -----Original Message----- From: "Peter Svensson"<psvasterisk@psv.nu> Sent: 01/02/05 16:55:52 To: "Asterisk Users Mailing List - Non-Commercial Discussion"<asterisk-users@lists.digium.com> Subject: Re: [Asterisk-Users] Zap channel occasionally misses dialing thefirst digit

Hi all, I have an asterisk box set up with a bri card (using zaphfc). I have a bri cellular gateway connected to it beacuse I'd like to route all my cellular calls through that gateway. The probel I encounter is that when trying to dial a phone number, I've the message : unable to create a zap channel. My card is normally well configured because when connected to the NT, It works