similar to: Intermittent Silence

I was doing a daily make update for asterisk. On the 19th the new version compiled fine. I installed it. All of my ata 186's can call out to pstn etc. All inbound calls, the phones ring but when you pickup, just silence both local and remote with no complaints in the cli. I backed down to the r 1.0 1.0.3 on the 20th CVS-v1-0-12/20/04-18:24:52 it worked ok. Yesterday I did a cvs update on the

I'm looking to expand my bandwidth for my Asterisk PBX. Why should I choose a T1 over DSL for my asterisk server? I found someone offering T1's for $290 a month + Loops or 3 Meg for $561 a month + Loops. Is this a good deal? Thanks Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus

My cdrs are missing accountcodes for incoming calls from other asterisk servers.. I've seen a few people mentioning this on the list and the solution seems to be setting up a dialplan for incoming calls from a particular sip peer.. in my opinion this does not scale well at all and I am looking for a solution to correct this problem. example sip peer: [asterisk_gw] type=friend

Completely new topic, and perhaps more important than it might appear just at a casual glance, if one hasn't been thinking about -- and really wanting to write -- some possible future (istic?) applications (as i have . . . and do). What i would like to have, which would _very_ much simplify what's involved in writing the code for certain things i have in mind, is (let me just say this and

Hi There, We currently using iproute2 for load balancing. However we need more upload speed as we load balance over 3 dsl lines. I''ve been looking for a way to combine the upload speed to more faster. Found a site called www.upstreaminter.net where these guys bond the adsl channels to improve uploads, Since downloading is problem as its need to know the ip address of the downloader they

Hi again, I keep posting about my problem with HTB -> http://mailman.ds9a.nl/pipermail/lartc/2005q3/016611.html With a bit of search I recently found the exact same problem I have in the 2004 archives with some graphs that explain it far better than I did -> http://mailman.ds9a.nl/pipermail/lartc/2004q4/014519.html and http://mailman.ds9a.nl/pipermail/lartc/2004q4/014568.html

Does anyone have any success using AudioCodes FXO terminating calls ? Ehsan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051002/5cfef736/attachment.htm

Hi, I looked at the docs and probably missed it: is there a way to set a dialplan on the GXP-2000? (to avoid having to press "Send") Thanks, -- "Computers are useless. They can only give answers." - Pablo Picasso

Hi, I've had an inquiry for a small UK call centre, mostly outbound calls. I get the impression they are mainly calling 3G mobile phones, monthly phone bill, with calls is approx ?5,000 for several feature lines. How feasible is something like this with asterisk? I guess one big question is which type of circuit to use, ADSL in the UK is only 256kbs upstream, some providers do bonding but

Our users are experiencing some unacceptable delay when trying to have a conversation. The delay is so noticeable that they keep stepping on each others words and resort to calling the customers via cell phone. Here is the setup SDSL Connection (PPPoA) Speedtouch 610 SDSL Modem 3Com 2224PWR Plus Switch (phones on separate VLAN) 8 Cisco 796 Phones All connecting to a remote Asterisk Server. We

I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial("IAX2/firefly@89280250/3",

I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get "you have" and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. -------------- next

Maybe I've missed it but I'm wondering if there has been any movement towards getting t.38 support into asterisk.. has there been any news? Where is t.38 support at? will it even happen? -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 279 bytes Desc: not available Url :

Haven't seen email since the 29th.. just testing. -------------- next part -------------- A non-text attachment was scrubbed... Name: mhess.vcf Type: text/x-vcard Size: 288 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050801/96555713/mhess.vcf

Hi there, Googling through the archives it looks like I'm the ferst person to want this... My aim is to set up a voicemail application with a custom greeting before *AND AFTER* the punter has left the message. Right now the relevant section of my dialplan is like this: exten => 2,1,Playback(/media/asterisk/answerphone-en) exten => 2,n,VoiceMail(2000,s) exten =>

Hi, I have one satellite link and one sdsl link. And, one LAN link. My load balancing is working great with a iptables''s masquerading (no fw rules). Now, I need to route the following ports to my sdsl link: - 20, 21, 22, 25, 80, 110, 143, and 443 Whereas the others go to my satellite link. Does anyone done this before? Regards, rootlinux __________________________________ Do

Hi List, I have a 2Mbps SDSL link which gets saturated during peak time because about I have about 3 E1 worth of g729 traffic going thru. So I'm planning to use IAX2 trunking to reduce bandwith requirement and squeeze out each and every bit of this (expensive) bandwith. I've set up two boxes (debian etch), one in a remote data center (which has plenty of bandwith) and one behing

Software zur Erkennung von "Spam" auf dem Rechner priamus.teamware-gmbh.de hat die eingegangene E-mail als m?gliche "Spam"-Nachricht identifiziert. Die urspr?ngliche Nachricht wurde an diesen Bericht angeh?ngt, so dass Sie sie anschauen k?nnen (falls es doch eine legitime E-Mail ist) oder ?hnliche unerw?nschte Nachrichten in Zukunft markieren k?nnen. Bei Fragen zu diesem

Hello all: I''ve read the documentation and am not quite sure where to start. What I''m trying to do is build a network with a 3 NIC Shorewall router. My system is behind a routed /49 network. I''d like to use 2 or 3 of the static IP addresses for my DMZ ( DNS server, mail, webserver etc ) and then have my remaining machines in a private network NATed is some way. My