similar to: Verizon SIP "trunking" Field Trial

Logan, How would I dump the object file generated by the JIT compiler pipeline? Could you point me to an example of how something like that is done? I’m used to working with the JIT machinery in memory but not writing object files out to disk. I’m have code to generate object files for AOT compilation - is it done the same way? Best, .Chris. On Apr 12, 2015, at 2:27 PM, Logan Chien

This is the only thing that I’ve found that works in terms of getting the exception to propagate out of the JITted function - change the “call” to an “invoke” and hook in the do-nothing landing-pad. https://gist.github.com/drmeister/7a35046f666826206973 Compare line 646 of the file above to the one that I just posted. > https://gist.github.com/drmeister/b97dec956c6ee9ffeb75 The first one

Logan, I need to make a correction of the post that preceded this one! I was wrong when I said this: > When I load the bitcode file for this module and then dump it just before it is JITted, the “uwtable” attribute has disappeared - is this prune-eh doing its work even though it’s not listed above in the list of function pass managers? The “uwtable” attribute does not disappear! I had

Hello, I use both ser/asterisk . In fact i wish asterisk to forward all the sip requests which are not handled by domain=domain.tld in sip.conf Here is a diagram: The sip agents use the Sip proxy as an outbound sip proxy and domain=domain.tld . When the sip agents dial sip:user@otherdomain.tld so the request is sent to sip proxy and so to Asterisk. I wish Asterisk to Look up the

Hi Christian, Thanks for your explanation. I know your situation now. I would suggest you to check the optimization pass used by the JIT compiler, especially IPO/PruneEH.cpp. It will try to add nounwind attribute to functions which will result in the problem you have mentioned earlier. Alternatively, as a workaround, try to add uwtable (function attribute) to the functions that are generated

I have a couple of questions about Digium's T1 cards, such as the TE410P. Any answers would be greatly appreciated. 1) Do they support standard T1s or are they ISDN-only? 2) Do you know of anyone offering support for configuring T1s for Digium cards, and if so at what cost? Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian

I am looking for estimated hardware requirements for running a 50 to 100 seat call center off of a single Asterisk server. The Asterisk server will have one quad T1 card installed (probably a Digium TE410P) with two T1s connected. The OS is Debian GNU/Linux (woody) with a custom 2.4.xx kernel installed. It is preferable for the server to have a single CPU and no shared IRQs. I would really

Is anyone succesfully running Asterisk behind verizon residential DSL? I seem to be having some problems with my Asterisk server switching to Verizon. I'm attempting to do some troubleshooting, but I'm really interested in knowing of anyone's setup that already has Asterisk working with Verizon residential DSL. Thanks AJ ------------------------------------------------------ This

Anyone out there have Verizon or SBC as the local exchange company in your area? I'm in the process of negotiations for additional interconnect agreements with these carriers (we're a wireless carrier). I'm looking for Asterisk folks who want a T1 (or several) for a PSTN gateway and would like to "network" their Asterisk boxes. This can take place at little or no cost

Hi, I'm having a problem with Verizon Wireless, I'm hoping someone here knows the right way to phrase the trouble report so it gets to someone at Verizon who can solve the problem. We have DIDs that simultaneously ring on voip lines, and Cell numbers. Verizon voicemail is turned off. Every thing works the way it's supposed to, UNLESS one of the cellphones is turned off, or in a

Hi, I HAD Verizon ONLINE. I was ABLE to mount samba served machines from the internet... eg \\strange.dreamhost.com\ from my XP box. I switched to Verizon AVENUE. Now I can't. (When I go through a VPN, though, I am able to mount the directory going around Verizon Avenue's network) I'm thinking this is a port restriction that Verizon Avenue has going on. I'm trying to

Hi, Has anyone on this list connected with Verizon's SIP product? We are currently undergoing interop testing with Verizon, and honestly, it seems like the most convoluted process. I'd be interested in talking with someone else who has gone through this and run a few things past you. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Is anyone out there using SIP Witch in conjunction with Asterisk? It claims to be able to "enhance existing IP-PBX solutions such as Asterisk", so maybe it can be used as a simple means to provide secure/encrypted calls. GNU SIP Witch - Summary <http://savannah.gnu.org/projects/sipwitch> GNU SIP Witch - GNU Telephony <http://www.gnutelephony.org/index.php/GNU_SIP_Witch>

I've dug down as far as I could on www.uspto.gov for anything remotely close to what is going on with Verizon and all searches end with only two possibilities in regards to what is going on. So unless the patent was issued to someone else and Verizon bought it, these are the only two possible patents this case could be based on... US 7,142,646 B2 Voice mail integration with instant

Greetings, A long time ago I setup a Linux machine as a Gateway/LAN Server using Verizon DSL as the ISP. I used the following HOWTO as the guide - DSL HOWTO For Linux: http://www.tldp.org/HOWTO/DSL-HOWTO/index.html Is there something comprable for Verizon FIOS? My Gateway machine runs Fedora. For a new server, I'm considering setting up a CentOS machine, while still using Fedora on my

Hi - We're moving up in the world to a PRI (Verizon), and I'm having some problems with it. I'm new to this PRI thing, though, so maybe I've just screwed up a simple config detail. I've got a TE410P on a Dell PE1600SC (ServerWorks Chipset). The card itself has a green light for the PRI, and Zttool shows Span1 as "OK". All I'm getting when I call into any

I am using Verizon FIOS to my home. I subscribe to a 5 MB down 2 MB up data package. I continue to pay for a standard voice line in addition to the broadband connection only for local calling, fax and emergency 911 use. The way it works is that the fiber optic connection is terminated on the house via an ONT (optical network termination). The ONT can provide connectivity for three types of

Apparently Verizon is blocking or changing packets on port 5060 so my softphone from my hotspot will not work. How do I set asterisk (11.23.0) to run default 5060 for all other devices I have - BUT for my software run on a different port like 5070? I'm using linphone and is easy to change the ports from 5060 to 5070 ( I think). Thanks, Jerry -------------- next part -------------- An HTML

Hi - Thanks for the fast response! > Try changing your dial statment to use a group instead such as > Dial(Zap/g2/3055551212) Same result: Spawn extension (from-sip, 000, 1) exited non-zero on 'SIP/2000-8dca' -- Executing Dial("SIP/2000-a4f1", "Zap/g1/19142246402") in new stack Dec 14 22:11:09 NOTICE[1506]: app_dial.c:800 dial_exec: Unable to create

Hello! We are open to the possibility of changing our business telephone number shortly. This will most likely be necessary due to a physical move, changing providers and a few other reasons. However, we woud like this to be the *last* time we need to do this. Ever. No matter what. Is that possible? On the Internet, you get this power with domain names. We "own" our domain