similar to: No voice from the callee

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

Hi friends, I am facing a problem with my asterisk 1.2 PBX. The problem is because of the CLI message "Remote UNIX Connection". After 2 days of a server reboot, this message starts coming. After it starts coming it still works well for few more hours, but then the asterisk hangs. During this time calls are still landing on the system, and calls are going forward upto the queue. Once

Hi, I use asterisk 1.8. My issue is : I have the same SIP members added to two queues. I use realtime configuration and has set the field ringinuse=0 for both the queues. But if an extension is answering the call in one queue, and some new call comes in the second queue, still that extension is ringed. In the queue_log table I am getting RINGNOANSWER events each second for the extension until

Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not

Hello, Let say Alice and Bob both have a sip phone connected to the same asterisk 11 box. Alice has T.38 enabled softphone. When Alice sends a fax to Bob extension, the following happens on my system: - Bob phone starts to ring - Bob answers - asterisk sends the incoming call to appropriate fax extension - Bob is hearing nothing at all: no tone, no sound at all. I want to play an audio file

Hi friends, Is there any way to prevent an Agent from logging in from a second extension if he is already logged on from an extension. Right now, the scenario is if he login from a second extension, asterisk will automatically log him off from first extension. What I need is that asterisk should tell him that he is already logged on from an extension and should prevent him from logging in

Hi friends, Do we have any way to prevent more than one Agent being logged in from the same extension? Also is there a way to limit an agent from logging in from more than one extension? I searched too much, but didn't find a solution. Please help. Thanks in advance. Shanavaz. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi In sparc ABI the arguments are saved by the callee in the caller stack frame. Q. What to do to save them in callee stack frame itself. for example by default this is generated sti r2, -2(fp) sti r3, -3(fp) Instead how to generate sti r2, 4(fp) sti r3, 5(fp) The Indices are just for explanation. The matter is In sparc they are -ive which means in the caller

hello, I use trixbox.I had define a feature code testfeature: [applicationmap] #include features_applicationmap_additional.conf testfeature => *3,callee,Macro,vote [featuremap] blindxfer => ## ; Blind Transfer disconnect => ** ; Disconnect Call automon => *1 ; One Touch Record atxfer => *2 ; Attended Xfer testfeature => *3 here is my macro-vote: [macro-vote] exten

Hello list, i am currently trying to implement tail call optimization in the X86 backend , so far i have it working for cases (modulo many unknown bugs :) where the tail called function is a destination within the source file and frame pointer elimination is performed. i implemented it as a dagcombiner transformation running in post legalized phase within the

Sent from my iPhone On Aug 8, 2007, at 10:46 AM, Arnold Schwaighofer <arnold.schwaighofer at gmail.com > wrote: > Hi Anton and Dale > first thanks for your answers. > > On 8 Aug 2007, at 16:43, Anton Korobeynikov wrote: > >> Hello, Arnold. >> >>> Is there a way to indicate that the register the tail call >>> instruction uses as destination

On Tue, Aug 9, 2011 at 6:19 PM, Ben Liblit <liblit at cs.wisc.edu> wrote: > I am using EQTDDataStructures (from the poolalloc project) to resolve > indirect function calls to over-approximated sets of possible callees. If I remember correctly, it only tries to resolve indirect calls. The analysis doesn't track direct calls because you can do it just as well yourself. Andrew

Hi, I've been playing with T.38. I observed that mostly but not always, it's the "calling endpoint" that reINVITE the other party to drop current SIP/G711 session and start a new T.38. But sometimes, it's also the callee party that reINVITE the calling party. Which is the "standardized" or most common, way to start a T.38 session ? Shall it come from callee or

Hello, you know , when a call setup, either caller hangup first or callee hangup first , the hangupcause will set to 16(means Call Clearing Causes) My question is how could i identify whether the caller or callee hangup the phone first? Best Regards! -- Thanks for your supporting, have a nice day. Sucan

You can send both paties to a meetme conference with Manager Redirect. Or if you are feeling more adventurous you could load the Manager Bridge patch that I posted to the bugtracker two months ago. It allows bridging of any two existing channels together through a manager action: http://bugs.digium.com/view.php?id=4297 MATT--- -----Original Message----- From: Roland Zagler

Hi, I have a requirement, which I am not sure whether it can be implemented. I had done some searches but didnt find an answer to this. Kindly let me know if some one has an idea to implement this: I have two Queues - Sales & Booking I have 12 Agents who are added to both the queues Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales Queue. Only 8 calls in the

Hello, I'm having problems with LIMIT_PLAYAUDIO_CALLEE in the Dial application. I want to play the limit file to both caller and callee at the same time, but it plays the limit file first to the caller and then to the callee. I searched the list and found someone with the same problem back in '06, but couldn't find any solution for the problem :( Anyone knows? Thanks, Best regards,

Hello - Apologies if this is not the right forum for this question. I'd appreciate any pointers. As background, I was trying to figure out a way to ensure that a function saves all the callee-save registers before executing, rather than just the ones it uses. I had guessed that marking them as clobbered would do this, but it turns out not to. Specifically, I would expect that on the x86-64,

I use ENUM lookup in my dialplan before sending calls through my PSTN trunk. One problem arises... When ENUMLOOKUP finds an SIP contact for that e164 number, Asterisk dials that contact, but when the remote server that should answer the call is down, or the IP link is down for some reason, the dial to PSTN trunk (which has the next priority) only takes place after the ring time of Dial

On Wed, Feb 08, 2017 at 01:00:24PM -0500, Waiman Long wrote: > It was found when running fio sequential write test with a XFS ramdisk > on a 2-socket x86-64 system, the %CPU times as reported by perf were > as follows: > > 71.27% 0.28% fio [k] down_write > 70.99% 0.01% fio [k] call_rwsem_down_write_failed > 69.43% 1.18% fio [k] rwsem_down_write_failed > 65.51%