similar to: Unknown warning messages

Hi all, With the gamut of FXO cards out there, I'm looking for a recommendation for home use. I have a nicely working Asterisk 1.4 system that just requires an FXO card to connect my NTL PSTN to it. My previous X101P clone seems to have kicked the bucket. Any suggestions would be greatly appreciated. Regards Kyle -- Kyle Gordon kyle@lodge.glasgownet.com http://lodge.glasgownet.com

Hey all, I'm running the latest and greatest CentOS4 here, along with Samba and Winbind coupled to an Active Directory server. It's all working smoothly, bar one little bit, the idmap gui and uid directives. It appears to be ignoring them completely. I've pasted the relevant directives below... winbind separator = + idmap uid = 10000-20000 idmap gid = 10000-20000 winbind enum users =

Hi all, Is anyone aware of any progress on this bug? http://bugs.digium.com/view.php?id=8763 Not only is the channel randomly disappearing during idle periods, it vanishes during a call as well. No indications in dmesg, syslog, asterisk or anything. Only cure is to rmmod and modprobe again. I'm currently on 1.4.0. Any ideas would be greatly appreciated. Cheers, Kyle -- Kyle Gordon

Hello Did anybody get any result for search in "Red Hat Hardware Catalog" http://bugzilla.redhat.com/hwcert/list.cgi?product=Red%20Hat%20Hardware%20Certification&quicksearch=3com&order=bugs.internal_whiteboard,bugs.bug_status%2Cbugs.priority%2Cmap_assigned_to.login_name%2Cbugs.bug_id&offset= On question for "3Com" I get only answer "15 certifications

Hi, I have * working with X-Lite and Sipura adapters, but I have one person who is linux based, and is trying to use Linphone and Kphone. His end works, but I get very bad echo on my end. Have any of you folks been able to get linux based soft phones working well with *? I'd appreciate links to howtos/docs if you have them, and/or samples of working configs for * and the linux

I'm experiencing a problem with a Sipura SPA-941 not available for incoming calls after Asterisk & Freepbx upgrade. I can dial out with the phone gto any other internal or external ext. It is registered with the Asterisk server. When I dial the Sipura directly from any other extension, it goes directly to vm. I have other Sip softphones that are working fine. A sip debug when calling the

The Arch Linux vcs packaging guidelines used to recommend shallow cloning of git packages, as apparently this was said to save bandwidth for the repo server. Now Pacman is doing the cloning automatically based on the vcs URL in the sources array, so the checkout function is no longer needed, and versioning is now done via a pkgver() function that in the case of git, uses git describe --long if the

All, I'm having an issue deploying softphones into my DUNDi/regcontext setup. My current design is that all SIP users get registered into a sipregistration context in the sip.conf. I then have a dialplan function that includes that and does the dial: include => sipregistration exten => _XXXX,2,Answer() exten => _XXXX,3,Wait(1) exten => _XXXX,4,NoOp(sipregistration call - Name:

In our experience, the total cost of softphones(money, reduced sound quality and lower reliability) in a large call center environment is actually greater over time than the cost of a channelbank and cheap analog headphones. We've tried 2 softphones, 2 kinds of SIP VOIP hardphones, 2 kinds of SIP analog adapters and we've tried channelbanks over the last 3 years. Right now we are half done

We are testing the new T38 passthrough code (SVN-oej-t38passthrough-r13347): - we are using a Sipura SPA-2100 as the T.38 user device - we are using a Patton SmartNode 2400 as the T.38/PRI gateway - we are using Asterisk in the middle We have the following in the [general] section of our sip.conf: t38pt_udptl = yes t38pt_rtp = yes When a fax call comes in from the SmartNode to Asterisk

Peter - 40 phones and only 3 PSTN trunks?. I would recommend at least 2 BRIs for this. If you have ISDN you can also get DDI to the extensions.I would strongly recommend abandoning the analogue PSTN lines and using ISDN. The setup pain you will go through will be significantly less, combined with better audio, more features (like DDI numbers!) and much more robust connections. You should look

Hi, As mentioned we have the problem that sometimes (could be up to a view times a day) for the calling party (SIP Device) you here ringing. The called party however answered the phone, but hears nothing. The calling party keeps ringing until the phone is hangup. First I thought maybe the card or the server has a problem, so I changed from a PCI beronet 4bri to a Junghanns 4bri PCIexpress and

Like everyone else who used asterisk with broadvoice, my outgoing calls died last week. I made the appropriate changes, and now basic incoming and outgoing calls are working. However, I have a few call-forwarding rules that are no longer working. It's certainly no coincidence. I can dial to all these number directly, but the problem only appears when there is an incoming broadvoice call, and I

Hi all! I am frustrated. I am new to asterisk. My system is ASTLINUX if receive a Fax on my sipura spa2000 i get this: Feb 25 07:41:00 NOTICE[1708]: rtp.c:564 ast_rtp_read: Unknown RTP codec 100 received -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060225/ca251876/attachment.htm

Jon, I actually had some more discussions with Tim on this issue, and it seems that the channel bank would still be a good option to choose for internal purposes. I would not see any other solution than a channel bank to connect many 2wire phones into one asterisk box. I had a talk today with Carrier Access, and it seems that the adit would do us fine. The fxs cards of the adit 600 are

Hello, I''ve added the field ''admin'' to my users database. When I did this, my unit tests broke with an error similar to this one for every test: ActiveRecord::StatementInvalid: Mysql::Error: Unknown column ''role'' in ''field list'': INSERT INTO users (`salt`, `hashed_password`, `role`, `id`, `login`, `email`) VALUES (1000,

I'm testing NVBackgroundDetect with Sipura-300 and I get this error: rtp.c:505 ast_rtp_read: Unknown RTP codec 100 received21 Does anybody know what is it? -- #Joseph

I have a SPA-2001 and I didn't realize I could use calling features on an analog handset. Does that mean you can dial *77 and use a VOIP feature? (like forward or hold)? Mike ________________________________ From: Jorge Carrasquillo [mailto:jorge.carrasquillo@gmail.com] Sent: Thursday, June 23, 2005 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Have you thought about making them agents, they would both be reachable by dialing there agent number then, and I know only one agent can be logged in at once. Just a thought. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Saturday, April 22, 2006 8:26 PM To:

For anyone who encounters this issue in the future, my issue was the configuration of llvm-gcc. Configure with --with-llvmgccdir=YOUR/PATH doesn't work, instead you are supposed to use --with-llvmgcc=PATH/TO/llvm-gcc and --with-llvmgxx=/PATH/TO/llvm-g++ That will at least get you past my error. Best, Kyle Quoting nicolas geoffray <nicolas.geoffray at gmail.com>: > I am