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...cannot access voice mail using phone or start recording while call is in progress and when i place a call to local sip extension there is a long pause ( 15 sec ) before the call gets dialled i assume that the problem would be due to the dial plan in PAP2 if so please help me changing it thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060307/a591bc4a/attachment.htm

HI I have two sip accounts from two different ITSP's both configured on asterisk server. how can i know if these accounts have been successfully registered ? i generally look at the /var/log/asterisk/full suggest me if there are better way of doing this thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060402/f2b1cf1c/attachment.htm

Hi I am looking at purchasing some DID lines from Teliax to install it on my asterisk. i would like to know some feed back on "Teliax" before i purchase. suggest me if there are better sevice providers. thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060330/98baf003/attachment.htm

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Hi, Created a zpool with 64k recordsize and enabled dedupe on it. zpool create -O recordsize=64k TestPool device1 zfs set dedup=on TestPool I copied files onto this pool over nfs from a windows client. Here is the output of zpool list Prompt:~# zpool list NAME SIZE ALLOC FREE CAP DEDUP HEALTH ALTROOT TestPool 696G 19.1G 677G 2% 1.13x ONLINE - When I ran a

Hi all, I'm setting up an IVR using Asterisk. Is there a way to have two streams played to the caller at the same time: for instance, one constant flow of background music, and the IVR contents at the same time? I've looked for solutions using (E)AGI and other things but nothing seems to work. Googling around and reading the list has not been helpful either... Thanks for your help,

Hi all. I am attempting to setup Asterisk to allow me to press *1 while in a call to use automon to record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. David Morrow

Helps if I send it to the list.... On Tue, Sep 1, 2009 at 5:33 PM, Giridhar S<thisisgiri at gmail.com> wrote: > Oprofile for Linux is a pretty good alternative. > (http://oprofile.sourceforge.net/about/) > > It uses hardware performance counters to collect profiling information > and therefore has very low overhead, whereas Valgrind performs dynamic &g...

Hello everybody, I am looking for a tool (in Linux or Windows) that allow me to get performance measures like cycle execution, cache accesses, etc. for an x86 architecture. I want to estimate the performance overhead due to the modification that I do using LLVM. Any suggestion is welcome. Thanks in advance, -- Juan Carlos -------------- next part -------------- An HTML attachment was

Hi, I'm speech recognition researcher and would like to do some research on recognition robustness in echo distortion of speech signal. Since VOIP is becoming wide spread, I'd like to simulate (one or more) common echo distortions that mostly appear in voip communications ? Any example, FIR or IIR filter or acoustical system response ? Any other distortion worth researching ? Thanks

...loaded a .wav file that has a welcome message . but when some one call the DID number it picksup the call and gives a long pause and then says "thank you " and hangs up. but if i press # key to access the directoy service it responds. please let me know what would be the problem thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060310/49f733d7/attachment.htm

Hi i am looking for a good ivr system for my company. these are my question are there any good ivr's that can be easily integrated with asterisk ? and are there any large scale deployment of asterisk to date ? thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060419/1a525818/attachment.htm

Got 1 issue I can't seem to knock out of this particular box. The IVR works fine on the zap channels and the incoming SIP routes. But coming in via the IAX2 route leaves me with a silent phone. The prompts all work still letting me navigate the menu. But just can't hear anything. This is with A@H 2.8 (Asterisk 1.2.7.1, with FreePBX 2.1.0 also installed) Any thoughts on where to

Today I converted my existing ext2 root partition to ext3. Since then my windows partion is not accessible in linux anymore. The mount command "mount /mnt/win" itself doesn't give any errors, but when I list the contents of windows partition ("ls /mnt/win"), I get the error "ls: /mnt/win: Not a directory". chdir also reports the same error. When I reboot the

I know this issue has been covered with at least 2 different patches, and probably a dozen different discussions, however I'm a bit unclear as to what my options are. I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall doing 1:1 NAT for machines behind the firewall. My asterisk box is one of these machines, and I'd like to allow foreign SIP clients

I am trying to get asterisk to email me my voicemail as attachments. What am I missing? Where do I tell it to go for SMTP services? Voicemail.conf: ; ; Voicemail Configuration ; [general] format=wav49|gsm|wav serveremail=pbx.agtcorp.local attach=yes maxmessage=180 skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 append=yes [default] 100 => 1234,Sean Garland,sean@siskiyoutech.com

List, The past few days the asterisk service on my server has crashed several times. I have had it running for months and have made no changes to it. When it crashes, I am unable to make calls or gain access to the CLI. The service has been stopped. If I try to start it again (service asterisk start), it will start and run for a few seconds then crash again. After a reboot, it will run

...o Formicuccia <giancarlo.formicuccia@gmail.com> Gianluca Anzolin <gianluca@sottospazio.it> Gianluca Gennari <gennarone@gmail.com> Gianni Antoniazzi <gianni.antoniazzi-ext@st.com> Giel van Schijndel <me@mortis.eu> Gilad Ben-Yossef <gilad@benyossef.com> Giridhar Malaval <giridhar.malavali@qlogic.com> Giridhar Malavali <giridhar.malavali@qlogic.com> Giridhar Maruthy <giridhar.m@samsung.com> Girish K S <girish.shivananjappa@linaro.org> Girish K S <ks.giri@samsung.com> Girish Verma <girish.gcet@gmail.com> Giusep...

...o Formicuccia <giancarlo.formicuccia@gmail.com> Gianluca Anzolin <gianluca@sottospazio.it> Gianluca Gennari <gennarone@gmail.com> Gianni Antoniazzi <gianni.antoniazzi-ext@st.com> Giel van Schijndel <me@mortis.eu> Gilad Ben-Yossef <gilad@benyossef.com> Giridhar Malaval <giridhar.malavali@qlogic.com> Giridhar Malavali <giridhar.malavali@qlogic.com> Giridhar Maruthy <giridhar.m@samsung.com> Girish K S <girish.shivananjappa@linaro.org> Girish K S <ks.giri@samsung.com> Girish Verma <girish.gcet@gmail.com> Giusep...

...o Formicuccia <giancarlo.formicuccia@gmail.com> Gianluca Anzolin <gianluca@sottospazio.it> Gianluca Gennari <gennarone@gmail.com> Gianni Antoniazzi <gianni.antoniazzi-ext@st.com> Giel van Schijndel <me@mortis.eu> Gilad Ben-Yossef <gilad@benyossef.com> Giridhar Malaval <giridhar.malavali@qlogic.com> Giridhar Malavali <giridhar.malavali@qlogic.com> Giridhar Maruthy <giridhar.m@samsung.com> Girish K S <girish.shivananjappa@linaro.org> Girish K S <ks.giri@samsung.com> Girish Verma <girish.gcet@gmail.com> Giusep...