similar to: [LLVMdev] machine pass

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

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

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

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

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 _________________________________________________________________

Hi, Looking for some reliable and quality providers of USA DIDs. Any pointers ? Thx Sans -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100623/816aecdd/attachment.htm

hi List and Manuel, I have encounter the following problem with the function "lineplot.CI".? I'm running R 2.10.1, sciplot 1.0-7 on Win XP.? It seems like it's a scoping issue, but I couldn't figure it out. Thanks! ...Tao > lineplot.CI(x.factor = dose, response = len, data = ToothGrowth)??? ## fine > lineplot.CI(x.factor = dose, response = len, data = ToothGrowth,

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, 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 All, I am newbie in this asterisk and a2billing technology . i had successfully installed asterisk in my server fedora -8 [server behind NAT/STUN] i after installation i can able to create users and tested the call features with X-Lite . the was working fine . after i installed the A2Billing in my same server with follow the steps from a2billing installation guide. but u cant access the

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

I have several remote phones that experience a slight "call" delay when answering phones, ie, they will answer, speak a few words, and then the remote caller will hear them, and the first half is cutoff? Any idea what could be causing this? Thanks, Bill. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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 need to perform some transformation after code was lowered. I think I need to do this in llc and use a MachinePass. However, it is unclear how. I am able to change the register allocator: llc -f -load XXX.so -regalloc=xxx foo.bc but I don't know how to just insert a different pass and pass new command options to llc. I found plenty of documentation about 'opt', but little

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 all, I have configured asterisk and a2billing.for inbound i have also configured did and its forwarded to sip extensions. But i want to enable queues with inbound numbers(DID).But i could not find a way to do this in a2billing. I want enable that if some did comes to asterisk/a2billing it should be forwarded to queues not sip extensions and their i want to enable hunting so if one

Hi everybody, This is the console output of the asterisk server. debian-te410*CLI> sip set debug peer 2002 SIP Debugging Enabled for IP: 172.26.48.113:5061 I have a sofphone with user 2002 registered on the server on the ip 113. I am trying to place a call to the sofphone on this ip. I have written a simple php script which utilises the exec_dial function inbuilt in phpagi.php file. I have