similar to: Can AMD detect Service Provider Message.

Displaying 20 results from an estimated 9000 matches similar to: "Can AMD detect Service Provider Message."

2007 Jun 04
4
Detecting card on the PCI Slot
I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. Regards, Sanjay Rajdev
2008 Mar 11
7
Best alternative for getting prompts recorded.
What is the best alternative for getting the IVR and other prompts recorded for Asterisk. Regards, Sanjay.
2007 Apr 16
6
BSNL caller ID (India)
Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683&nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev
2009 Jul 02
3
[LLVMdev] LLVM under Syllable
I tried to Build LLVM under Syllable, but it was failed on the next moment http://clip2net.com/clip/m0/1246547164-clip-99kb.png any ideas? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.llvm.org/pipermail/llvm-dev/attachments/20090702/346df366/attachment.html>
2007 Mar 29
5
SIP RTP Tunnel
Hello, is it possible to rout ALL RTP Data over Asterisk, like SIP1 <---RTP---> Asterisk <---RTP---> SIP2 I know it seems quite useless. But I want to simulate a IAX -> SIP connection and have no Phonecard installed on my computer ;) Thanx, Kalle
2007 Mar 29
2
Problem while using asterisk Realtime
I am having problem while having asterisk work with ODBC (Postgres) The error that I am getting is "config.c: Realtime mapping for 'sippeers' found to engine 'odbc', but the engine is not available" I really donot know what has went wrong. I have set the ODBC connection properly I have verified it using :: [root@asterisk ~]# echo "select 1 " | isql asterisk
2008 Apr 03
4
C# SIP API to Comiunicate with Asterisk
Do anyone has an idea about an open source SIP API written in C# that can communicate with Asterisk, to call out? Regards, Sanjay.
2007 Apr 13
3
LED does not glow on new Voicemail
I have a CISCO 7912 phone, the LED on the phone does not glow when there is new voicemail, can we configure Asterisk to have the LED glow on new Voicemail. Regards, Sanjay Rajdev
2007 Apr 11
3
missing chan_zap.so
Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0. All SIP accounts were working fine, today I tried to install a fxs Sangoma A200 card and got the following error. [Apr 12 01:15:17] WARNING[31018]: channel.c:3024 ast_request: No channel type registered for 'Zap' [Apr 12 01:15:17] WARNING[31018]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'Zap'
2008 Feb 15
2
Voice activity detection
Hey sorry to hijack this thread, but I just remembered a request I wanted to make to the speex devs. I tried using the activity detector, but I just couldn't get it working well. I ended up using my own, where I think it just considered voice on if it passed a certain threshold (I know, pretty primitive). I also tried one that checked for a signal, like if the strongest frequency
2004 Feb 11
6
Spelling (PR#6570)
I came across this in connection with an unrelated issue > beta[2] Error in beta[2] : object is not subsettable > beta[2] <- 5 Error in "[<-"(`*tmp*`, 2, value = 5) : object is not subsetable One of the messages must be wrong, but I need a native English speaker to tell me which one. -- O__ ---- Peter Dalgaard Blegdamsvej 3 c/ /'_ --- Dept. of
2008 Mar 14
2
Logs for Call generated by Manager API
I am generating an outbound call through the Manager API and bridging it to an internal Extension, my problem is I am not able to find the logs for the call generated by the Manger API, Since on the same Asterisk server there are many users connected and I am receiving lot of Events back, not able to recognize which was the call generated by me as same time multiple users are dialing out.
2016 Jan 27
2
asterisk 13 mixmonitor - random missing syllables
hi, i have strange problem with asterisk 13 mixmonitor, recording to wav (centos6) when the system is under load, there are sometimes missing syllable there arent BIG spikes on cpus recordings are to ramdisk (/dev/shm) any hints? -- --------------------------------------- Marek Cervenka =======================================
2016 Jan 27
2
asterisk 13 mixmonitor - random missing syllables
Dne 27.1.2016 v 13:14 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> hi, >> >> i have strange problem with asterisk 13 mixmonitor, recording to wav >> (centos6) >> when the system is under load, there are sometimes missing syllable >> >> there arent BIG spikes on cpus >> recordings are to ramdisk (/dev/shm) >>
2010 Aug 25
3
What does this warning message (from optim function) mean?
Hi R users, I am trying to use the optim function to maximize a likelihood funciton, and I got the following warning messages. Could anyone explain to me what messege 31 means exactly? Is it a cause for concern? Since the value of convergence turns out to be zero, it means that the converging is successful, right? So can I assume that the parameter estimates generated thereafter are reliable MLE
2007 Apr 03
2
Require only GSM Codec
Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev
2016 Jan 28
2
asterisk 13 mixmonitor - random missing syllables
Dne 27.1.2016 v 17:50 A J Stiles napsal(a): > On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >> Dne 27.1.2016 v 13:14 A J Stiles napsal(a): >>> On Wednesday 27 Jan 2016, Marek ?ervenka wrote: >>>> hi, >>>> >>>> i have strange problem with asterisk 13 mixmonitor, recording to wav >>>> (centos6) >>>> when the system is
2008 Mar 28
1
how to register IAX user without password for any user
Dear Sanjay, Sorry sanjay i miss to explain completely. My PC2PC mean is Dialer2Dialer i want to allow call between Dialer with out any registry and authentication through IAX. so i need to setup Asterisk accept calls from any user and users can call to each other without any password and registration. please help how can i configure Asterisk using IAX in this regards. thanks, Asif Message: 9
2003 Jun 30
3
if else statements in R
Hi, there I am a grad student struggling to get my syntax right in a model I am building in R. I am trying to specify a variable,W, in a loop, such that on the first iteration, it is equal to Wo, my starting value, which I have already specified with Wo<-9.2 On any other iteration other than the first, I want W to equal the previous value of W in the iteration. plus an increment from
2007 Mar 20
1
centos raid 1 question
Hi, im having this on my screen and dmesg im not sure if this is an error message. btw im using centos 4.4 with 2 x 200GB PATA drives. md: md0: sync done. RAID1 conf printout: --- wd:2 rd:2 disk 0, wo:0, o:1, dev:hda2 disk 1, wo:0, o:1, dev:hdc2 md: delaying resync of md5 until md3 has finished resync (they share one or more physical units) md: syncing RAID array md5 md: minimum _guaranteed_