similar to: SPA-2002 call HANGUP. May be a SIP bug.

[I'm sorry to trouble the list with this, but this is the only way I know to contact the person concerned] This message is for Stephen Karrington - it appears that you have over-agressive 'spam' filters and we can no longer email you. Please rectify this if we are to have meaningful conversation! The original message was received from Linus Surguy

Hello, I am looking for a VOIP Service Provider to work with getting started with Asterisk. Does anyone have any brief recommendations? Ease of use and support are the key criteria. ----------------------------------------------------------------- Ed Mansouri Ucompass - http://www.ucompass.com Make sure we stay connected to you Add yourself to the Ucompass Address Book

Hello! Asterisk box receiving calls. Is there some way to get information about current calls from external or AGI application? I'm interested in: - duration, how long calls already in the system (billing and actual time); - source/destination phone numbers; - etc. In other words can I receive information which we are usually getting in CDRs during the time when the call is still active?

Is there an easy way to get information about local calling areas out of telcos? I'm trying to get a list of area codes and prefixes in my local calling area out of Verizon, and it looks like they've stopped providing the information online. Is there an easy source that I'm missing, or do I need to call them and have them mail me a copy every few months? Scott

Morning Asterikians, I've just got my nice shiny quadBRI card, and it seems to be working very well - except for one little issue - CallerID. The card is currently connected to an ISDN2e line in P2P mode, and an S0 adapter on our existing alcatel PBX. The S0 connection recieves callerID and displays it correctly - the 2e line doesn't, and BT have said that CLID was enabled on the

We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. We have a flash frontend thats tied to our backend mysql DB. We use it for loading web site traffic data, email opens, click-throughs, bouncebacks, stats, etc. It could also be used with

Hi, Hopefully this won''t be too off-topic (I''ve seen both bonding & vlan mentioned on the list, but not really together). I''ve tried to get bonding (2 x 100Mb EEPro, but will want to try on 1000BaseT) and vlans to work together, but without luck. I can get them working fine (seemingly at least - I didn''t tried bursting on the bonded port) individually.

On your web you have a link http://www.voip-info.org/wiki-Asterisk+settings+Diamondcard To Setup Calling with Diamondcard.us and I signed up and paid the money according to Stephen Karrington it was all automated... And it was automated to take money but when you look for service hookups or information you don't get it. I have tried now for last little while to contact them for support

Hello Everybody, We need to open some ports on the target machine, let''s say 694:udp and 3306:tcp I couldn''t find any documentation how can I do this with Puppet. What the best way of doing this? Thank you Dmytro -- You received this message because you are subscribed to the Google Groups "Puppet Users" group. To post to this group, send email to

Hello! I've got the latest asterisk sources from cvs dated Jan 26 2004. And was trying to build it under FreeBSD 4.8 Main part was built fine. I face problems only while compiling h323 channel. Its Makefile has hardcoded linux libs (openh323 and pwlib) and some defines. Also -lpthread was replaced with -pthread and -ldl with -dl. So I've got successfully build and installed asterisk.

hello, I am trying to find where filtering is done in order to drop packets randomly there (eg.10%of them) for a testing purpose in a school project. I want to do this after packet is filtered and before being sent. I d aprreciate if you could help or suggest. Thanks.

Hello, list members, can i limit pps rate with linux? How? -m limit does not fit, as i understood: it can help with low rates only (is that true? any suggestions?) Thank you, -- _,-=._ /|_/| `-.} `=._,.-=-._., @ @._, `._ _,-. ) _,.-'' ` G.m-"^m`m'' Dmytro O. Redchuk

Hello, Our company currently stores its data on centralized samba servers which causes performance problems for remote office employees. We are thinking of decentralizing the data with NAS in each office and synchronizing all the NAS on our central domain controller with MSDFS. I have heard this can be done in real time with Windows and Active Directory.

Hi, currently I''m using 48 class with htb & very stable Is there any maximum number of class I can create in a single linux box ? I need 500 or even 1000 class for campuss network. Any help appreciated thanks & regards Tino _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

hello, i have a question: - which one is faster: "tc filter with u32 match per dst ip" or "iptables match per dst ip with target CLASSIFY"? - this question is for large rulesets (over 500) thank you, cristian carstea

Folks, I've been following recent discussions regarding echo and echo training with much interest, since it's a problem we've never been able to eliminate here. We're facing two challenges presently, and they may be related (or not): a) Cisco 7960s in the office here echo back to our staff, but the customer hears decent sound. b) We've yet to be able to pass faxes through

Hello Everybody, I have configured a Linux box that does traffic shaping. Its working wonderfully fine, just as expected...Now i want to measure the bandwidth consumed by each of my hosts....But I dont want SNMP to run on all the hosts[as required by MRTG] I used iptraf on my linux box, but it only measures the bandwidth on interface basis only.... I tried installing traffic-vis, its not working

Sorry if this was posted yesterday, I was having issues with being auto-unsubscribed because of my spam filter. Not sure if my post made it through. Hi everyone, I'm wondering if I'm missing something obvious here, or if Asterisk just doesn't support what I'm trying to do. It seems like it should be simple, but appearances can be deceiving. I've got an Asterisk box

Greetings, I have one PSTN line connected to my Asterisk@ Home box with call waiting. I also have an SPA-2002 connected to an analog phone. When I am calling on the PSTN and a call waiting beep comes through, I can hear it, but when I press the flash key, nothing happens. It is as if the Sipura is not passing the flash through. I monitor the asterisk box with the verbosity turned up, but

Hi all, updated py-htbstat: http://www2.ldc.net/~dor/py-htbstat/ Some screenshots: http://www2.ldc.net/~dor/py-htbstat/shots/shots.html It''s now possible to use classids in form ''xxxx:xxxx'' now (actually i dont know why i''ve made it to use ''1:xxxx'' only :-) -- _,-=._ /|_/| `-.} `=._,.-=-._., @ @._, `._ _,-.