search for: altcall

...le of weeks now and find that it's extremely slow for configuring. For instance, it takes several minutes to boot up, apply any changes via the web interface takes at least a minute, etc. Is this normal behaviour? Is there anything that can be done about it? Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

...terisk says "that starts with 99. let's strip off the 99 and call 751234 at FWD, IE: sip:751234@fwd.pulver.com:5060". Is that possible, or would services such as FWD reject the call since the device making the call (Asterisk) hasn't registered? Thanks! -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

Hey guys. If my Asterisk box connects to the PSTN using SIP and IP over ethernet and doesn't require any authentication, what sort of a trunk would need to be created? Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

...Auto destroying call '0fbefde17f0c2a7a168e1d3f158e5760@127.0.0.1' What am I doing wrong? I noticed that Asterisk said that the verbosity level is >= 19, but the debug message that appeared should've been suppressed by ``sip no debug'', no? Cheers, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

Hey Everyone, We are in the process of reviewing headsets for use with our GXP-2000s. I'm looking for some feedback as to which headsets people are using, the pros and cons of those headsets, and if they would recommend them to someone else. Any help would be appreciated... - Jason -------------- next part -------------- An HTML attachment was scrubbed... URL:

...formation is up-to-date. Many of the examples suggest using insecure=very for the SER entry in sip.conf, but I believe that option is for pre-1.2. If you would care to share which settings work best for adding an entry for SER into sip.conf, that'd be great. Thanks, -- Nick e: nick.hoffman@altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

Hello all, I want to sniff all these info to test a sip ip phone talking to a asterisk server. I have used tcpdump, but It just shows the UDP, length: 602 Anyway to see the sip uri. Host info? Regards, Dinesh. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

Does anyone know if a TDM2400 will fit into a Dell 2850? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - <mailto:kerryg@techdatapros.com> kerryg@techdatapros.com <http://www.techdatapros.com/> http://www.techdatapros.com -------------- next part -------------- An HTML attachment was scrubbed...

Hi all I think this could be en old question. I would like to do a realtime billing prepaid system, mainly using asterisk. I have found few things; I can not get CDR function into agi because asterisk set them once the call is absolutely finish (at least main values for the main porpouse, billsec,duration, etc..) There is a patch that allow you to use CDR

Looks like I am going to be doing my first serious commercial install of FreePBX. I DO mean serious. They are willing to put up with a few glitches initially which is why I have decided they will be a good first client. I have turned down several over the past couple years because I just did not feel comfortable with the state of software/hardware. It seems to work much better now. I was

Hi there, I've seen people using SER with Asterisk. I took a look at SER website, and I didn't see the point in using it, since Asterisk already handles SIP very well (apparently, at least). But, as I'm starting, and some of you (more experienced) use it, I know that there's something there... So I would like to know why to use SER. Is it because of scalability, performance,

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

Hi, When ever I put g729 in allow for trunk the other two codecs (ulaw and alaw) stop working and I get the frame type error for them, but g729 works fine. I've cleared general part of sip.conf of codec info to be on safe side. If ulaw and alaw are the only ones allowed they work fine. Asterisk shouldn't be doing any encoding or decoding, all codecs should be passing through. Any

Hi When I make a call with the Grandstream GXP-2000 through Asterisk (and SER) to landline using VSP, after I hang up the call the other party are still connected for another 30-40 seconds. I've notice that the SIP BYE is sent to Asterisk, but Asterisk sends no SIP BYE on to VSP. When I use the SPA-941 the call terminates on the other right away soon as I hang up. I have updated the

Hi, I have Grandstream GXP-2000 connected to Asterisk, and Asterisk has trunk to VSP for PSTN calls. When ever I place local PSTN call, the landline doesn't hang up right away (40 sec), when I hang up the GXP-2000. The GXP-2000 seems to have problems making international calls as well. Where it hangs up soon as the other party picks up. I have used different IP phones, VSP's and etc.