similar to: [LLVMdev] passes after lowering‏

Greetings List,Trying to interconnect with a new provider.. the require a?compliance?with?RFC 3261 ?so knowing less than needed about RFC documentations.. i would like to know if Asterisk is actually in compliance with?RFC 3261 or not..?Can any one help with this? Regards -- Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP USA: +1 347 562 2308

Hi, I am instrumenting certain calls, and want to add an extra argument. Say original: foo(int x, int y) changed into modified: foo(int x, int y, int EXTRA) This is in opt, before lowering. Given the list of original arguments, is it possible to tell the stack offset of the EXTRA argument? Thank you, Dan _________________________________________________________________

Not sure if this is a bug yet, so I wanted to ask around to see if anyone else was having this issue. I have pollmailboxes=yes set in voicemail.conf but externnotify is not called. I know it isn't the externnotify script because if the changes are done in asterisk then it is called properly, if the changes are done via our webserver then it is not. Also, we use odbc voicemail storage. Thanks

Hi, I have a situation where the files I'm backing up are written to every fifteen minutes or so. There's a good possibility that rsync will try to copy a file while it is being written into, and I'm wondering if there's any risk that the _source_ file will be damaged? I've seen similar posts about the target file, but my concern is the source. Additionally, would the writing

Hi, LLVM documentation is not clear. Is it possible to write a machine pass? I am trying to insert some machine code before the return instruction. ideally, I'd like a pass that runs the last one before generating assembly. How can this be done? Thank you, Dan _________________________________________________________________ Hotmail has tools for the New Busy. Search, chat and

Greetings List. I'm facing a strange issue with one of my providers.. after sending an INVITE request my server places the call on hold.. until the call is answered.. this is happening only with this provide although i have 3 other providers i route calls through.. can anyone explain what is going on? -- Tarek Sawah / Integrated Digital Systems / CCNA, MCSE, RHCE, VoIP / +1 347 562 2308

Sorry if that's a vague subject, but this problem is a little weird and I'm just wondering if there are any suggestions out there. We've got a Samba server (3.0.23) running on a CentOS 5.3 server offering up a data share of 7TB on an XFS filesystem. The authentication all happens through a Samba PDC with an LDAP backend all on a different server. The system in question is just a

Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _________________________________________________________________ The

Hi, What kind of C/C++ high level code can generate a computed jump, such as: jmpq *%r14 or jmpq *(%r14,%rbx,8) ? I imagine that any calls (including virtual) would use something like 'call *%r14', and the above jumps are mostly from 'switch' statements. Is this correct? Anything else? Thank you, Dan

Hi, I'm trying to access a share on my work network using smbclient. We have an Windows Active Directory network. My client computer is running Solaris 10 u8. The computer hosting the share says it's running Acopia ARX(3.0.0b1) According to Active Directory (not familiar with this OS, i think it is a NAS) I run this command to get the Kerberos ticket. bash-3.00$ kinit jtmb at

I am running rsync in cygwin on windows. I am attempting to backup a somewhat large data store (750GB) to a remote site. As its windows and preserving permissions exactly is important, I have an iSCSI drive mounted on the local system across a somewhat slow WAN link (IE, it would take about 3 months to copy the datastore over it). Unfortunately, since this appears as a "local" copy to

Greetings list,this question is rather a pain in my side.. i have been trying to figure it out.. it could be simple.i have a customer with a callcenter .. we developed a CRM "Customer Relations Management" with an SIP dialers built in.the question is the following.. is it possible to force the agents (users) to use a certain UserAgent which is the one built-in our system? this way will

can anyone know where i can find information on compile hmisc on windows, especially 64 windows? thanks, _________________________________________________________________ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. ID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 [[alternative HTML version deleted]]

hi, i'm looking for documentation on configuring asterisk to work with a modem that should work with an analog line. i don't see the info in the handbook or reference manual or o'reilly's. any references and/or links, much appreciated. thanks. g. _________________________________________________________________ The New Busy is not the old busy. Search, chat

what is the general view about the versions of the packages that are used with asterisk. lame asterisk asterisk-addons dahdi libpri i like to say on a version and not upgrade due to my experience with Linux and upgrading screwing up things. When it comes to Asterisk i have only one server build under my belt and I have had issue along the way. What do most people do with the software

i have this card installed Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) following the steps below found on freepbx site cd /usr/src/dahdi-linux-complete-2.2.0.2+2.2.0 make make install make config /sbin/ztcfg echo "/sbin/ztcfg" >> /etc/rc.d/rc.local cd /usr/src/libpri-1.4.10.2 make clean make make install when i run make config i do not get

here is the issue phones freepbx-2.7.0]# ./start_asterisk start STARTING ASTERISK Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. mpg123: no process killed ----------------------------------------------------- Asterisk could not start! Use 'tail

Hello, On the Samba3 Release Planning wiki page, Samba 3.2 is designated as "discontinued". There is a statement: "As this strategy is quite new, we are currently still providing security releases for 3.2" Anyone know how long 3.2 will continue to receive security releases? Thank you.Alex http://wiki.samba.org/index.php/Samba3_Release_Planning

Is there a setting or easy way I can remove the code that makes R only listen for web traffic on 127.0.0.1? I've noticed it seems to reject traffic to both the regular help files (i.e. what starts up when I do "?function"), as well as if I try to connect to pages I've made for the R.rsp package. We could live without the help files, but it would be neat to allow others to access

This is the problem: Call coming into a queue in ringall strategy, if a member (SIP) of the queue is busy when entering the queue, and this member comes free after a little time, the member never rings.. How to solve this? I changed all parameters of the queue with no results... Wath i need: If one member of the queue is busy when a new call come in to the queue, this member can hangup and