Re: [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?
Hi Bruce, On 01/03/2011 06:03 AM, Bruce B wrote: Thanks for the input. The errors I pointed out should be available in the /var/log/messages or messages.1 .2 .3 etc...or in dmesg each time you reboot the machine. I have just rebooted the server and can confirm that neither dmesg, /var/log/messages or /var/log/syslog contain device not accepting address or USB device is disconnected error messages. Sebastian At least that is the case with mine. If you don't have it then it's I guess very specific to this atom board. I know someone else added something along the lines irq=fixconflict to get rid of the issue with a wireless USB mouse showing same symptoms and CentOS. IRQ conflicts is also one of my thoughts as well but I can't play at BIOS level at this time as the server is thousands miles away from me. Regards, On Sun, Jan 2, 2011 at 8:18 PM, Sebastian s...@open-t.co.uk mailto:s...@open-t.co.uk wrote: Hi, On 01/02/2011 08:08 PM, Bruce B wrote: Thanks for the input. I have the latest drivers but it seems that there is some serious incompatibility issue with the kernel as when the FLASHING happens even if the system is restarted it's still not detected. One has to re-plug it in and then it shows in wanrouter hwprobe. It could also be that the atom board is not compatible with the driver in my case. It your /var/log/messages upon restart do you any line that might match this: * * *cd /var/log/* *grep -o device not accepting address ** *grep -o USB device is disconnected ** *dmesg | egrep device not accepting address * *dmesg | egrep USB device is disconnected * If not then you are not experiencing the same issue. If you have then it's a universal issue and not hardware specific. I would really appreciate it if you look into your logs and let me know. If you are asking me to look into the logs when the U100 is not recognised and it is flashing - I'm afraid I can't help much - as this problem went away and haven't had issues any more for at least 6-8 months now. Considering the U100 persist in giving problems even after computer restarts somehow sounds to me like a hardware issue. I wouldn't be surprised if the Acer Aspire motherboard is not coping very well with both of them being plugged in - although, obviously, it should. This linux-usb FAQ seems to also suggest that the problem is hardware (based on the device is not accepting address error message): http://www.linux-usb.org/FAQ.html#ts6 Have you tried disabling in bios everything you don't use - maybe sound card, wifi etc. to free some IRQ's? I run kernel 2.6.33.4-smp - just in case it helps. Let me know if you need any other info. If you find a final solution - I would be curious to know - just in case I will stumble over this problem. It wouldn't be a bad idea to build a script which grep's lsusb output - and run it through cron maybe every 10 minutes - to try and determine more precisely when does the U100 stop being recognised. Then you can look through the logs to see if there is anything else happening in the system around the same time. Sebastian Regards, Bruce On Sat, Jan 1, 2011 at 6:57 PM, Sebastian s...@open-t.co.uk mailto:s...@open-t.co.uk mailto:s...@open-t.co.uk mailto:s...@open-t.co.uk wrote: Hi Bruce, On 12/28/2010 10:51 PM, Bruce B wrote: Thanks for the input. I can not replicate the situation as it happens randomely or maybe over the weekend. However I have sent you all the requested command and logs in a separate e-mail for your analyzes. The only thing that stood out at me was the output of lsusb -v at the very end where it timed out. Since all lines didn't work I am to assume that both module went down but per my diagnoses with hwprobe I could see one unit connected and the other was not when the problem happened. Simply connecting/disconnecting that unit or connecting it to another port solved the problem and it showed up in hwprobe This is an Acer Aspire Revo mini PC. I am wondering if the U100s draw too much power? The only other USB connected device is the thumb size wireless connector for the keyboard. Acer computer: http://reviews.cnet.com/desktops/acer-aspire-revo-ar1600/4505-3118_7-33777218.html Don't know if this will help - but I will
Re: [asterisk-users] Saving the monitor file on new file always using Monitor(wav, Record1, m)
Hi,
On 01/01/2011 05:43 PM, bilal ghayyad wrote:
Dear List;
For each call (in specific case), I need to do a record and save in a spearated
file, so I am thinking the best thing is to save based on the time.
Monitor(wav,Record1,m)
So, how can I make the file name to be based on the current time (which is
changed always, or based on the some unique paramter (related to the call it
self).
I use something like this in extensions.conf for outgoing calls:
exten =
_9.,1,Set(REC_DIR_OUT=/shares/phone_calls/${STRFTIME(${EPOCH},,%Y-%m-%d)}/outgoing)
exten = _9.,n,Set(REC_FILE_OUT=${STRFTIME(${EPOCH},,%Y-%m-%d %H %M %S)}
- ${EXTEN:1}.gsm)
exten = _9.,n,System(mkdir -p ${REC_DIR_OUT})
exten = _9.,n,MixMonitor(${REC_DIR_OUT}/${REC_FILE_OUT},b)
exten = _9.,n,Dial(SIP/${EXTEN:1...@my_voip_provider)
exten = _9.,n,HangUp()
Sorry for the line breaks. My email client does that. You should keep
each extension priority on a single line.
This will create one variable for the folder (containing the date today)
and for the file (containing the time of the call and the number
dialled) - and then creates the folder and starts MixMonitor with the
filename as argument. If your setup is larger, you should also add maybe
the calling extension to the file name - so that you don't have two
files with the same name - if two extensions try to call the same
external number at exactly the same time (seems unlikely to me).
Sebastian
Any advise?
Regards
Bilal
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Re: [asterisk-users] Log and forward calls to cellphone?
Le 01/01/2011 18:32, Gilles a écrit :
On Wed, 29 Dec 2010 16:55:46 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
I wouldn't be one of your friend: when I'm calling you I call a landline
but finally will be charged for a mobile call (imagine I have free calls
to landlines from my ISP). I give you an information: in France you
don't have the right to do this unless you have it precise *before*
redirection.
I checked with the VOSP: Apparently, it doesn't support getting an SIP
message to forward calls on the fly, and I pay for the forwarded leg
of the call (the caller will pay his part).
As you are a Free Telecom customer, why not using your freephonie
account to forward incoming calls to your mobile?
Something like in you POTS incoming context:
...
exten =
s,n,Dial(SIP/${Phone1}SIP/{MobilePhoneConnectedWithWIFI}IAX2/${SoftPhone},21,${DIAL_OPTIONS})
exten = s,n,Goto(s-${DIALSTATUS},1)
exten =
s-NOANSWER,1,Dial(SIP/freephonie/${MyMobileNumber},30,${DIAL_OPTIONS})
exten = s-NOANSWER,n,Hangup
exten = s-ANSWER,1,Hangup
exten =_s-.,1,Voicemail();other cases
--
Daniel
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[asterisk-users] digim tdm2400p fxo fake answer supervision problem.
Hi. I am using Digium TDM2400PFXO with Asterisk1.6. When call sets in the box , it answers the call even the phone is not picked. ideally it should answer the call when the phone is picked up. Its charging the clients. Please let me know how can I cover this ? Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digim tdm2400p fxo fake answer supervision problem.
On 01/03/2011 06:16 AM, Muhammad Usman wrote: Hi. I am using Digium TDM2400PFXO with Asterisk1.6. When call sets in the box , it answers the call even the phone is not picked. ideally it should answer the call when the phone is picked up. Its charging the clients. Please let me know how can I cover this ? Thanks in advance. If you already know the internal extension you are going to dial when a particular FXO port starts ringing, you do not need to answer the inbound call before calling Dial() in your dialplan. The Dial() application will automatically answer the originating call when the called party answers before bridging. Now if you don't know which internal extension you're going to dial you are out of luck. You'll have to answer in order to get the DTMF digits of the extension from the caller. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback form to place on site for customers. Recomendation to achieve this.
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R Sent: Sunday, January 02, 2011 3:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Callback form to place on site for customers. Recomendation to achieve this. On Sun, Jan 2, 2011 at 1:04 AM, JP CR jprollersk...@hotmail.com wrote: I want to place a form on my site so customers can recieve an mmediate callback and the PBX should connect them to a cell sales agent. Are there anfree modules available for this, or one should code this from scratch? What I want is when a potential client submits his number... the PBX dials the number makes an announcement and dials an extension (which is actually a cellhopne dahdi member) and makes the connection. You could look at http://phono.com for the PhonoSDK. It can connect you to skype, sip or PSTN destinations via Tropo applications. I wonder though if/how you are going to have a way to check if the number given isn't a joke. What prevents people from just typing in all kinds of phone numbers at 6AM in California or something? Is anyone working on that problem? Do you restrict it to hours that are safe all over the USA? Is there a check of area codes? Color me just curious... /r I use the Asterisk::Manager module in PERL to do just this sort of thing. If you did the same thing using Apache and PERL you could address each of these issues. I'm sure there are PHP or other ways to do this, PERL is just my development standard due to other issues. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] changed datadir
I am trying to configure mysql to use a different datadir than default in order to move this to a larger volume. I have copied all mysql data from /var/lib/mysql to my new volume and ran both chown -R mysql:mysql * and chmod -R 660 * in order to setup correct ownership and rights for the data. It was working for a few days until today upon going into mysql and typing show databases, I receive the following error: ERROR 1018 (HY000): Can't read dir of '.' (errno: 24). I quickly double checked my rights and found them to be correct, then I restarted mysqld and this has fixed it for now. Has anyone else run into this problem? I am concerned that it will happen again if the actual cause is not corrected. I am running the latest Fedora on new quad core Dell hardware with 24GB RAM and 1.7TB of disk. Any help and/or suggestions much appreciated. Thanks. Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] changed datadir
Sorry, sent this to wrong group list. On Mon, Jan 3, 2011 at 10:13 AM, Nicholas Hart nh...@partsauthority.comwrote: I am trying to configure mysql to use a different datadir than default in order to move this to a larger volume. I have copied all mysql data from /var/lib/mysql to my new volume and ran both chown -R mysql:mysql * and chmod -R 660 * in order to setup correct ownership and rights for the data. It was working for a few days until today upon going into mysql and typing show databases, I receive the following error: ERROR 1018 (HY000): Can't read dir of '.' (errno: 24). I quickly double checked my rights and found them to be correct, then I restarted mysqld and this has fixed it for now. Has anyone else run into this problem? I am concerned that it will happen again if the actual cause is not corrected. I am running the latest Fedora on new quad core Dell hardware with 24GB RAM and 1.7TB of disk. Any help and/or suggestions much appreciated. Thanks. Nick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoIP PoE phones for restaurant
Greetings, I mailed the list regarding an intercom system some months ago and we came to the conclusion that I should purchase a Polycom 501 phone. I'm now considering the purchase for this year, and I'm now wondering between the Polycom 501 and the 320 for the intercom. I don't need the spare ethernet on the phone because I would like to have my voice network separate from my regular LAN. Which one would be easier to use, the 501 or the 320? I want PoE, were these both made before PoE was standardized and do I need a special cable? Can I make this cable myself? In the future we plan to have 7 phones in the house. I'm considering what kind of PoE switch I should purchase. I have 3 PoE access points (for two separate LANs). I've been considering th HP ProCurve 2610-24/12PWR Switch (J9086A) ( http://h10010.www1.hp.com/wwpc/il/en/sm/WF06b/12883-12883-3445275-427605-427605-3751584-3658873.html ) It's got 12 PoE ports, it's managed, and it looks like I can pick one up for under $500. Any help is appreciated. -Andy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callback form to place on site for customers. Recomendation to achieve this.
On Sun, Jan 02, 2011 at 12:04:07AM +, JP CR wrote: What I want is when a potential client submits his number... the PBX dials the number makes an announcement and dials an extension (which is actually a cellhopne dahdi member) and makes the connection. You might try something based on this: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out It's easy to generate a call file which dials the agent's phone, waits for a pickup, and then dials out. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue announce parameter problem
Any idea ??
Em 28/12/2010 11:32, Eduardo Lobo Blanco escreveu:
Hi list,
I´m using Asterisk 1.6.2.14.
I have some queues configured in my solution, using dynamics members.
All my members are IAX2 clients.
Each member will be at least in two queues at same time.
So i´m trying to use the announce parameter in queues.conf to inform
the member
from where this call arrived to him.
But it is not working. The queue in general is working fine, but the
audio for annouce is not played,
and i can´t see any error message about that, like missing file, codec
problem, etc ...
Here is my queues.conf:
[general]
persistentmembers=yes
autofill=yes
monitor-type=MixMonitor
shared_lastcall=no
[queue_template](!)
musicclass=default
strategy=random
joinempty=yes
leavewhenempty=no
ringinuse=no
monitor-format=gsm
timeout=20
retry=1
maxlen=5
[client_SP](queue_template)
announce=queue_client_sp
[client_RJ](queue_template)
announce=queue_client_rj
[client_PR](queue_template)
announce=queue_client_sp
[client_SC](queue_template)
announce=queue_client_sp
[client_RG](queue_template)
announce=queue_client_sp
In extensions.conf :
[entry-queue]
exten =
s,1,Set(MONITOR_FILENAME=/var/log/asterisk/calls/in/${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)}_${queue}_${CALLERID(num)})
exten = s,n,Queue(${queue}65)
Any idea why Asterisk dosnt´play the announce ??
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[asterisk-users] Clarification on DAHDI Fax Detection
Hi folks, I was hoping that someone might be able to help clarify some confusion I have on DAHDI Fax detection after spending some time searching. My understanding is this: 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. 2.) faxdetect=incoming will, upon detection of a CNG tone, send the call to the fax extension. 3.) faxdetect=outgoing will ?? Also, do Digium cards with HW Echo Cancellation detect the CNG tones in hardware? If so, how does the faxdetect setting in DAHDI affect that behavior? Many thanks, Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On 01/03/2011 11:26 AM, Tom Rymes wrote: Hi folks, I was hoping that someone might be able to help clarify some confusion I have on DAHDI Fax detection after spending some time searching. My understanding is this: I'll try. 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. No. CNG tone is never used to affect the state of an echo canceller. All G.168 compliant echo cancellers will respond to the CED tone (generated by the answering endpoint) and will reconfigure the echo canceller appropriately. Most modern ECs will *not* be disabled, but will enter a 'linear' mode where they can do some echo suppression but not complete cancellation. DAHDI will detect CED when most software echo cancellers are in use and will disable them (since none of the available software ECs can go into linear mode). The Digium HPEC software EC will detect CED on its own and enter linear mode. 2.) faxdetect=incoming will, upon detection of a CNG tone, send the call to the fax extension. If the CNG tone arrives from the network side of the DAHDI channel (the far endpoint), then yes. 3.) faxdetect=outgoing will ?? The same thing, but if the CNG tone is being sent towards the DAHDI channel (from the near endpoint). This is rarely used. Also, do Digium cards with HW Echo Cancellation detect the CNG tones in hardware? If so, how does the faxdetect setting in DAHDI affect that behavior? No, none of the Digium HW ECs detect and report CNG tones via the DSP; CNG tone detection is still done on the host CPU. 'faxdetect' is not set in DAHDI, it's set in chan_dahdi.conf. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma U100 failing every Monday - USB port problem or Wanrouter issue?
On 01/03/2011 11:20 AM, Bruce B wrote: Thanks a lot for that Sebastian. I will report back my findings when I find the resolution on this. I'm a bit late here, but I can say that Sangoma support has always been extremely helpful when I call them. If you haven't already, definitely give them a call. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote: On 01/03/2011 11:26 AM, Tom Rymes wrote: [snip] 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. No. CNG tone is never used to affect the state of an echo canceller. All G.168 compliant echo cancellers will respond to the CED tone (generated by the answering endpoint) and will reconfigure the echo canceller appropriately. Most modern ECs will *not* be disabled, but will enter a 'linear' mode where they can do some echo suppression but not complete cancellation. DAHDI will detect CED when most software echo cancellers are in use and will disable them (since none of the available software ECs can go into linear mode). The Digium HPEC software EC will detect CED on its own and enter linear mode. OK. Either way, though, the changes to echo cancellation are not affected by the faxdetect setting, right? 2.) faxdetect=incoming will, upon detection of a CNG tone, send the call to the fax extension. If the CNG tone arrives from the network side of the DAHDI channel (the far endpoint), then yes. Great. This is the typical usage, I presume, directing fax machines to FFA, Hylafax, another fax machine, or hangup (if this isn't a fax line). Is there a time limit to when DAHDI listens for faxes (say the first 10 seconds of a call?), or might it detect one in the middle of a ten minute call? 3.) faxdetect=outgoing will ?? The same thing, but if the CNG tone is being sent towards the DAHDI channel (from the near endpoint). This is rarely used. [snip] I figured that must be it. Presumedly you might use this to perform some activity on an outgoing fax prior to sending it, such as logging something, etc? Maybe send it to FFA, receive it, and e-mail it to another server that faxes it out on a local number to save toll calls, etc? Thanks for the clarification, there's a lot of conflicting info out there. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoIP PoE phones for restaurant
Andy, The 501 and 320 are EOL. I'd go for the IP335 and a 2626-PWR, since the 2626 can support vlans you can isolate data and voice. Make sure to spec a UPS on the PoE switch. On Mon, Jan 3, 2011 at 8:30 AM, Andy Graybeal andy.grayb...@casanueva.comwrote: Greetings, I mailed the list regarding an intercom system some months ago and we came to the conclusion that I should purchase a Polycom 501 phone. I'm now considering the purchase for this year, and I'm now wondering between the Polycom 501 and the 320 for the intercom. I don't need the spare ethernet on the phone because I would like to have my voice network separate from my regular LAN. Which one would be easier to use, the 501 or the 320? I want PoE, were these both made before PoE was standardized and do I need a special cable? Can I make this cable myself? In the future we plan to have 7 phones in the house. I'm considering what kind of PoE switch I should purchase. I have 3 PoE access points (for two separate LANs). I've been considering th HP ProCurve 2610-24/12PWR Switch (J9086A) ( http://h10010.www1.hp.com/wwpc/il/en/sm/WF06b/12883-12883-3445275-427605-427605-3751584-3658873.html) It's got 12 PoE ports, it's managed, and it looks like I can pick one up for under $500. Any help is appreciated. -Andy -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Clarification on DAHDI Fax Detection
On 01/04/2011 04:22 AM, Kevin P. Fleming wrote: On 01/03/2011 11:26 AM, Tom Rymes wrote: Hi folks, I was hoping that someone might be able to help clarify some confusion I have on DAHDI Fax detection after spending some time searching. My understanding is this: I'll try. 1.) Echo cancellation is automatically disabled upon recognition of a CNG tone, regardless of the faxdetect setting. This can only be disabled at compile time. No. CNG tone is never used to affect the state of an echo canceller. All G.168 compliant echo cancellers will respond to the CED tone (generated by the answering endpoint) and will reconfigure the echo canceller appropriately. Most modern ECs will *not* be disabled, but will enter a 'linear' mode where they can do some echo suppression but not complete cancellation. DAHDI will detect CED when most software echo cancellers are in use and will disable them (since none of the available software ECs can go into linear mode). The Digium HPEC software EC will detect CED on its own and enter linear mode. That's not true. Modern echo cancellers normally disable completely. It is arguable whether they should disable completely for FAX, but they need to behave properly for all modems. For any duplex modem, disabling only the NLP is useless. They need to cancel end to end, so they don't get upset by a continuously adapting canceller, and so they can minimise the issues caused by the highly non-linear G.711 channel. 2.) faxdetect=incoming will, upon detection of a CNG tone, send the call to the fax extension. If the CNG tone arrives from the network side of the DAHDI channel (the far endpoint), then yes. 3.) faxdetect=outgoing will ?? The same thing, but if the CNG tone is being sent towards the DAHDI channel (from the near endpoint). This is rarely used. Also, do Digium cards with HW Echo Cancellation detect the CNG tones in hardware? If so, how does the faxdetect setting in DAHDI affect that behavior? No, none of the Digium HW ECs detect and report CNG tones via the DSP; CNG tone detection is still done on the host CPU. 'faxdetect' is not set in DAHDI, it's set in chan_dahdi.conf. Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Replacing digital pri card
I don't imagine this would be too complicated - don't have any experience with AsteriskNOW - but on a 'vanilla' linux distro it would just be a matter of making sure dahdi is loading the correct drivers and doing a couple of minor config file updates. On Tue, Dec 28, 2010 at 3:01 PM, Tyler Davis tda...@zulily.com wrote: I need to replace our current 1 port pri card with a quad port card. I'm currently using the newest AsteriskNOW distro. Are there any issues I should expect to run into? I'm hoping the transition will be smooth, however I havent had to do this in the past. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 1.8 MIBs
Cannot find asterisk-mib.txt and digium-mib.txt anywhere. Were they dropped? -kkm -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question About Conferencing Capabilities
My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Freeswitch). I've been poring through the docs, and have even set up a test server myself, but there are some very specific things we are looking for that I can't figure out if Asterisk can do or not. We want to be able to do the following: - Create dynamic, on-the-fly conferences that can remain active even when initiating user leaves - Within a conference, give users the ability to mute and/or deaf individual users - Give users the ability to enter a whisper mode with another user - where they are holding a private conversation that can only be heard by the two of them ( It sounds like the Meetme module has a functionality like this, but it is a little vague in the documentation) - Allow users to be in two conferences at once; the user would most likely have one muted at any given time so as to hear the other one, but we want them to be able to switch back and forth easily Could anyone advise me on whether Asterisk can accomplish these needs, or perhaps what it might take to do so? We are not averse to doing some customization if we can find the people who know how to make it happen! Thanks, Siobhan Hamilton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question About Conferencing Capabilities
My company is building a VOIP application, and initially were just using a barebones OpenSIPS implementation to host one-on-one calls; however, we want to expand the functionality to conferencing (which, of course, OpenSIPS doesn't handle) and was looking into Asterisk (the other option being Freeswitch). I've been poring through the docs, and have even set up a test server myself, but there are some very specific things we are looking for that I can't figure out if Asterisk can do or not. We want to be able to do the following: - Create dynamic, on-the-fly conferences that can remain active even when initiating user leaves - Within a conference, give users the ability to mute and/or deaf individual users - Give users the ability to enter a whisper mode with another user - where they are holding a private conversation that can only be heard by the two of them ( It sounds like the Meetme module has a functionality like this, but it is a little vague in the documentation) - Allow users to be in two conferences at once; the user would most likely have one muted at any given time so as to hear the other one, but we want them to be able to switch back and forth easily Could anyone advise me on whether Asterisk can accomplish these needs, or perhaps what it might take to do so? We are not averse to doing some customization if we can find the people who know how to make it happen! Thanks, Siobhan Hamilton -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1.8 MIBs
On Mon, Jan 03, 2011 at 08:04:48PM -0800, Kirill Katsnelson wrote: Cannot find asterisk-mib.txt and digium-mib.txt anywhere. Were they dropped? They're now part of the wiki. https://wiki.asterisk.org/wiki/display/AST/Asterisk+MIB+Definitions https://wiki.asterisk.org/wiki/display/AST/Digium+MIB+Definitions -- Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
