similar to: Réf.: RE: SIPP Load testing

And the whole idea of using an enum service is to save those costs and also to encourage intelligent use, If I'm going to call Duane to ask him a question I'm going to call him on his mobile if he's not at home hang the expense but the point is that is wasted money that us intelligent people do not need to spend in the first place. Cheers, Dean -----Original Message----- From:

e164.org is a public name service which provides ENUM.164, a method devised by the IETF and ITU to allow an ordinary telephone to be connected to an Internet type network and provided dialling service from other, regular telephones. Unlike many other "free" voice over IP systems, e164.org allows users who have a regular telephone line, to also hook themselves up to the Internet

For those interested, I'm giving a talk about VoIP/enum.164/asterisk tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS build #2, 4th floor, room 10. Sorry for the late notice, it didn't occur to me that there might be people on this list interested and able to attend etc... -- Best regards, Duane http://www.cacert.org - Free Security Certificates

I'm running the latest version of cvs (not stable), I'm not sure what the other end is running and if this has been fixed or not yet, however I was playing round with onhold earlier, the call went to onhold, and came back from it, then 2 seconds later was hung up unexpectedly, below is what was on console... -- Started music on hold, class 'default', on

Firstly we've setup a SIP proxy that uses e164.org to do enum lookups, also rather then issuing people with yet more numbers they have to remember we've coded up a watered down version of e164.org for people that would just like to have a single SIP phone rather then run their own PABX. http://www.Like2Fone.Com for more details on that service. The plan is to get people hooked on VoIP

Soren Rathje wrote: > Next thing will probably be a sbl.e164.org service to block spammers like we do with email... :-) Actually we don't need to do that, using normal NAPTR record can be used instead. We know the IP the call is coming from, we can find out from the NAPTR where calls normally go to based on the phone number, if the 2 don't match filter it. -- Best regards, Duane

Long time coming, but we finally have a 3rd party interface on the website to add block of enum numbers in regex form... eg +4412345[678] which will match +44123456 +44123457 +44123458 also +4412345[16-18] which will match +441234516 +441234517 +441234518 or just short prefixes +4412345 so anything starting with +4412345 will match... Currently this is accessible via web interface

Stefan Gofferje wrote: > Hi folks, > > I would like to have kinda catchall function for incoming sip > connections. A channel, where everybody could connect to by dialling the > url e.g. sip://guest@<myserver>, like the [guest] section in iax.conf. > I have played around a bit but any attempt to dial any > extension@<myserver> without prior registration leads to a

There is now a peering arrangement between e164.org and FreeWorldDialup which means any and all subscribers on FWD are now easily able to make enum calls by prefixing their call with **164, like wise it's almost as simple to make a call to FWD by hitting 8829990<fwd number> This means that for those of you wanting to send/receive calls to/from FWD subscribers you can now do so, easily

For a long time now we've allowed people to publish a wide variety of URI against their enum records such as SIP/IAX2/H323 for VoIP and other types for non-VoIP such as HTTP/MAILTO etc. For the most part these record types aren't listed or aren't utilised so I've done up a quick hack for firefox users as a proof of concept and I'm hoping others will take advantage of this and

Hi, sipp (http://sipp.sourceforge.net/) seems to be a good app. Take a look at http://www.voip-info.org/wiki-SIPP on the wiki to have more info about it... Basically, there is scenario which are describe there and I personnally generated about 3,000,000 calls before having to restart asterisk and i placed about 90 concurrent calls. Good luck! -----asterisk-users-admin@lists.digium.com a ?crit :

Can you put this patch on line? (I don't think it's too big...) In my mind, the main objective is to create a special field and force its value in chan_capi.c and check wether it goes through asterisk or not... What do you think of that? Regards ---------------------- > >jean-marie.goupil@telintrans.fr wrote: >> OK, so I'll do that... Is there any infos I need to know

right, so far, here is what I've done: I succeed in take in a new variable the UUS1 field sent with the connection request for incoming calls. It was quite simple afterall... (I just had to find where the data CMSG->Useruserdata is coming in chan_capi.c) Now I would like to know where this field is instanciated for outgoing calls in order to control this step? I am looking for that but I

I tried that but it still doesn't work... I think I don't have the correct approach. Have I to install any ISDN drivers (=modules ?) BEFORE dealing with CAPI ? If yes, why shouldn't I use the hisax drivers (which are kernel ones) instead of fcpci drivers (which doesn't seems to work, by the way...) And finally, how is it possible to link the two modules together? As you can see,

Steven P. Donegan wrote: > I don't want to be negative here, but I do believe people who go to do this know the potential risks they face. In many countries (4 of which I have direct, although several year old experience with - all in Asia) taking a local phone line and attaching asterisk to it and gatewaying traffic from other countries will be considered to be 'theft' by the

Hi, I'd like to test Asterisk performance under more concurrent sip calls. I use Sipp, but do get "Unexpected message for Call-ID ...", so I wonder if anyone is using sipp succesfully with Asterisk and is willing to share more info about his solution ... Any other convenient way to load test Asterisk ? Is sipp the right tool ? Thanks in advance, regards, Rob. sipp: The

Grandstream v1.0.4.68 firmware http://www.hellofone.com/downloads.html Seems to have loaded ok on my BT100.. -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to

Hello, I'm stressing an Asterisk 11 platform with incoming calls from sipp 3.1. I've dedicated a context to sipp in my dialplan. Everything works OK expect that calls from sipp comes in with a CallerID set to sipp and this sipp value is stored in CDR. 1. I can change the value of the CallerID but how can I have the calls from sipp traced in CDR with a customized src field value ?

Hello everyone, Recently we've been focusing quite heavily on making Adhearsion[0] faster. To do that, we needed a convenient way to test our Asterisk voice apps. The obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a little clumsy to use sometimes, especially if you're trying to use it to drive interactive calls like an IVR. So to make our own lives

We are in the process of implementing realtime voicemail. I was wanting to "stress-test" the system to see if or when it would fall over. Is it possible to use sipp to create say 250 calls, each of which leaves a message in the voicemail ? My dialplan is currently [default] exten => stress,1,Answer() exten => stress,2(vm),Voicemail(7777|su) exten => stress,3,Hangup()