search for: jhiver

Hi List, I have set up the following in my extensions.conf ; local numbers look like 0262XXXXXX ; but must be dialed 262 262XXXXXX exten => _0262XXXXXX,1,Dial,IAX2/543@voipjet/011262262${EXTEN:4} exten => _0262XXXXXX,2,Dial,IAX2/jhiver@NuFone/011262262${EXTEN:4} exten => _0262XXXXXX,3,Congestion It did work for a while, however when dialing I get: stargate*CLI> -- Executing Dial("SIP/2000-a939", "IAX2/543@voipjet/011262262914202") in new stack -- Called 543@voipjet/011262262914202 Dec 1 13:2...

...outside asterisk but I couldn't resist. I wrote just a little perl toy I wrote to find words in existing phone numbers. For example, using the script on Digium's toll free line, you discover that it can be 877-LINUX-ME but also 8775-GNU-WOE or 8775-HOT-ZOE ;-) Here's the output: [jhiver@fly jhiver]$ perl detect-strings.pl 8775468963 GNU (468) matches 8775468963 GO (46) matches 8775468963 GOT (468) matches 8775468963 HOT (468) matches 8775468963 IN (46) matches 8775468963 IO (46) matches 8775468963 JIM (546) matches 8775468963 KIM (546) matches 8775468963 KIN (546) matches 87754689...

Hi List, I am working on least cost routing code on the moment, and I am stumbling on a problem. Say you have provider A having: Prefix XXX 0.10 Prefix XXXYYY 0.20 And provider B having Prefix XXX 0.15 You're stuck, because you cannot decide if provider B's "XXX" prefix also covers XXXYYY numbers or not. If it doesn't, it would be a waste

Hi List, I've had some issues with some VoIP providers where either: 1 - There is massive PDD but finally the call goes through 2 - There is massive PDD but the call gets rejected anyways I was wondering if there was a way to automatically detect such behaviors when it happens (maybe with a script or something) so I can take the faulty providers out of the routing and maybe automatically

...Capable of translating prefixes so that you can generate dialplans using different dialing locales (for example, using french dialing conventions along with US providers) I'm sure that they are many quirks and bugs to sort out. You can check the documentation here: http://search.cpan.org/~jhiver/Asterisk-LCR-0.03/lib/Asterisk/LCR.pm Installation amounts to: perl -MCPAN -e 'install Asterisk::LCR' Let me know how it goes! Cheers, Jean-Michel.

How to set-up LCR ? a. which companies can be used with LCR? b. how to set-up & maintain LCR? c. multiple connection to one gateway? Example: +886223456789 could be reachable via a. ENUM free b. Dundi free c. Voipstunt free d. Voipbuster free e. Nufone $ f. Voipstunt $ g. others with 4 concurrent connections $$ h. others with 3 concurrent connections $$ I am looking

...to who-knows-where. Amazingly, it wasn't getting tangled. To me, this was emblematic of how tradition dies hard... T.30 will be with us for a while to come. Wise managers will limit this to the outer boundaries of their enterprises wherever practical, ASAP. From: Jean-Michel Hiver <jhiver@ykoz.net> Doing a analog (piece of paper) -> digital (scanning process) -> analog (modulation over TDM) -> digital (conversion to TDMoIP) -> analog (demodulating on the other fax) -> digital (reconstructing the image in fax memory) -> analog (printing) conversion doesn'...

Hi List, I am looking for a 1 FXO analog termination device, other than the obvious PC + FXO card, and which can achieve decent call quality. The SPA-3000 seems an option... have you got any other ideas? Cheers, Jean-Michel.

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

Hi List, Yesterday night after a power off due to a faulty UPS my asterisk doesn't want to start anymore. Here is what I get on the CLI: Asterisk Ready. *CLI> Disconnected from Asterisk server: Bad file descriptor. Executing last minute cleanups == Destroying musiconhold processes Asterisk uncleanly ending (0). I use 1.2.7 I think on a debian sarge and cdr_pgsql too. Any ideas?

We're getting our feet more and more wet with VOIP at work. We want to experiment with a good wireless (as in WiFi) phone. What would be a good phone to impress my boss with? I'm personally drooling over the UTStarcom F3000, but compatibility and shipping ETA info is a bit sketchy. Phil

While reading about how to maximize capabilities in asterisk i have read about SER and OpenSER. The sites do not explain to newbies (maybe that's on purpose) what are the benefits of using those products tied with asterisk (or is SER an asterisk replacement??) Can someone give me an idea of what's the usage for open(ser) and asterisk? is it for scalability? should I run it in the same

Hi List, Being very new to Xen I have a few generic questions for the list, I hope to grab some useful advice and pointers to documentation. I am evaluating Xen to consolidate a few existing servers into one appliance (mainly in order to reduce power consumption, heat, and hardware failure risks). I plan to have a SER router, an Asterisk LCR router, a voicemail server, a calling card server

Hi Everybody, Got a weird problem here I think. Got a setup with an asterisk (current from cvs as of a few hours ago) in a box with an el-cheapo ISDN BRI card connected to the PSTN network and two Snom phones internally (one Snom-100 and one Snom-200). Dialing between the snom phones or dialing out to PSTN from any of the snom phones works perfectly. But when I receive a call FROM the PSTN

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Hi List, I have a test asterisk box with a TDM400P with 4 FXO modules plugged in. Yesterday I could use the box without any issues - no problems. This morning, the sound on the box was absolutely horrible. After some fiddling about, I have rebooted the box, and now asterisk refuses to start! Here's the message I get: Jun 9 10:45:53 WARNING[3297]: chan_zap.c:769 zt_open: Unable to

Hi List, I have slapped together a no-frills yet functional prepaid framework for Asterisk. It supports concurrent calls and has been built with robustness, simplicity and billing accuracy in mind. You can find some docs and the code on the following page: http://ykoz.net/intl/grobill/ It's licensed under the GPL. If you have any questions feel free to email me directly. Best Regards,

Hi List, Have you got experience with this product? http://www.voipsupply.com/product_info.php?products_id=885 From its description, it looks like the ideal appliance to set up some "double play" ISP data / telephony offer and I was wondering if anybody was using it and what it was worth. Cheers, Jean-Michel.

Hi, Asterisk::AGI is a fantastic piece of software. Unfortunately it comes with NO LICENSE WHATSOEVER. That's very annoying when you want to write GPL stuff that depends on it. I have tried mailing the author some time ago with no response. Does anybody know what the software license for this module is? Cheers, Jean-Michel.

According to the wiki, "IAX2 jitter buffer (when turned on) doesn't currently work well with trunking (trunk=yes in iax.conf) Yourcall...sstartto....soundlike....this :-)" Is this still the case? If that's so, how do you do to use trunking in conjunction with a PRI card? Do you use another, separate asterisk box for the jitter buffer? Cheers, Jean-Michel.