similar to: UDP Ports scan on firewall

Hi, I have access to two providers. On one of them the authuser is the same as the username, so outgoing works. On the other one I can only get incoming - what ever combination I try for outgoing I get an error. The register command has the ability to specify both usernames (which is why incoming works) but outgoing doesn't seem to, and without that I'm stuck. They are defined as:

Hi Tony, Try adding "fromuser=xxxxx", maybe "username=xxxx" isn't enough... Just a guess, it already solved a few problems for me. -Manuel -----Messaggio originale----- Da: Tony Hoyle [mailto:tmh@nodomain.org] Inviato: martedì, 18. maggio 2004 13:03 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Configure asterisk for outgoing.. need authuser parameter?

Has anyone out there used the PipeMedia/PipeCall PSTN gateway? Anything good/bad to say about it? I'm considering using them for a new customer. They seem to have good rates, good provisioning tools and (better still) give commission on usage to dealers. -- David Gurr Congruity Ltd. Fax: 0871 661 1756 Hemel Hempstead UK

Any recommendations for a VoIP provider in the UK? I have a few guys in a field office in the UK with SIP phones and a VPN tunnel back to a working Asterisk setup in the US. The Asterisk setup has an IAX trunk with TelaSIP/VoipXpress with local DiD's for US offices, so they can call vendors, customers etc in the US at local rates. I'd like to get the same thing for the UK, so that UK

There was some discussion on this list recently about the voiptalk silver service. I've just had an e-mail from them saying that the price has been reduced to 2.99 per month. However, they still only provide an 0870 number whereas pipecall provide a local call rate 0845 number in the fee. Chris

I have been a reseller & subscriber of pipecall since they started, however I am really struggling to get pipecall to work for outbound or inbound calls. I get errors that the registration has timed out. I have tried many variations of the register command register => 0845xxxxxxx@sipproxy.pipecall.com/1000 register => sipxxxxxxxxx:xxxxxxxxxx@sipproxy.pipecall.com/1000

Still no luck using asterisk as a proxy. 48 hours solid working on this. I'm beginning to think asterisk isn't going to be compatible with the provider I'm using :( Has anyone got *any* clues as to what can cause this message? It's definately provider specific (voiptalk works, pipecall doesn't) but confusingly seems to be caused by something in the client phone app. I

Hi everyone, I want to get multiple DDI's and hunting across those DDI's in case one of the lines is busy using analog phone lines. The system is for a large house so I want 3 x PSTN lines. 3 x DDI's and the ability for those DDI's to be presented across all three PSTN lines. BT say you can't have more than one DDI number associated with a PSTN line and that you can't

You should be able to get ISDN though. I think BT provide this anywhere. Otherwise your solution will be a Primary rate (ISDN30) with 8 channels. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Matt Sent: 07 June 2004 10:19 To: asterisk-users@lists.digium.com Subject: Re: RE: [Asterisk-Users] Multiple DDI &

On Thu, 17 Feb 2005 15:04:50 +0100 Stefan Gofferje <stefan@gofferje.homelinux.org> wrote: > Hi folks, > > I'm registered with sipgate, a German SIP provider. >Configs works fine so far. Trouble is, after a while, it >seems, my registration is dropped by sipgate. How do I >tell * the interval for * registering with a provider? I >suppose, the re-registration

I've configured voicemail.conf to delete voice mails after they are emailed. The email work ok but the message don't delete. The config is as follows: [default] 3000 >= 1111,CS,cs@x.com,,delete=yes Thanks, Chris

I would like asterisk to dial an extension or external number but for the call to only be connected after the called party presses a key. Therefore been able to announce the call to the called party before answering. I have had this working on queued calls but want to incorporate this for standard dialled extensions. Our use for this would to be able to divert a call to a users mobile but only

>> So it sounds like it should "just work". I'll let you know in a few >> weeks time :) > TDM400P to E1/T1 card faxing fails by design. The lack of > synchronisation between cards means it can *never* work with any > reliability. The hardware will not permit it. >> >> (And I know I've asked this before, but any recommendations for a >>

Hello, sometime back, I saw an article (i think it was in powerpoint) comparing Asterisk for suitability for call centres. This article has features on one side and tick for each feature that * has. I cant seem to find that article. Infact I have no idea where I saw it except it was on the internet. I have searched on Google but come out blank. Need someone's help on it. Can someone point me

Hi all, I've been playing about with Asterisk for years now on and off, just SIP to SIP calls, using FWD and suchlike. I'm moving house at the beginning of September and have decided to build an Asterisk based system for my home office. I'm in the UK and wonder if anyone can give me advice on lines and hardware to use. Had planned to go with an ISDN2e line coupled with a BT

Hi, At the moment, libpri /w Asterisk 1.6, Dahdi 2.1, is not supporting NT-Point to multi-point mode. Here (France), most small PBXes are connected to ISDN through BRI trunks in PtmP (don't know why but it seems the general case). So this NT-PtmP function would be very helpful to easily slide an Asterisk box between an existing PBX and the network. Does the same case apply elsewhere (UK,

Hi hpa, the following patch corrects the usage of '__likely', which has been replaced by 'likely()' in recent kernels. Plus it adds a missing include 'klibc/compiler.h' for sys/types.h as the bitops on S/390 insist on using 'likely'. Please apply. THX, Hannes -- Dr. Hannes Reinecke hare@suse.de SuSE Linux AG S390 & zSeries Maxfeldstra?e 5 +49 911

Volodya, This program is in error. Either the format specifier needs to be %lld or the argument value X needs to be casted to type long. This will compile fine but it probably won't work at runtime because printf is expecting a long, not a long long. I think your backend producing "wrong results" is actually correct :) Most modern C/C++ compilers will detect this for you at

Folks, I have completed a final draft of the bytecode documentation. You can find it in llvm/docs/BytecodeFormat.html in CVS. Or, if you'd prefer to view online, you can find it here: http://llvm.x10sys.com/llvm/docs/BytecodeFormat.html I'm soliciting your feedback on this document. Please let me know if anything is awkward or just plain wrong. Thanks, Reid. -------------- next part

What can I do because my X100P don´t answer sometimes. I put these parameters in the zapata.conf, but sometimes my X100P don´t answer. usedistinctiveringdetection=yes dring1=363,137,0 dring1context=incoming2 Any sugestion? Thank´s --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! -------------- next part -------------- An HTML attachment