[asterisk-users] Calls from asterisk
When we have calls that originate click-to-daial apps that use the manager interface they always originate from asterisk is there any way to change the from name? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 chan_h323, help please...
Hi, My configuration is SipPhone--*1---*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)--asterisk1---(H323)--asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best h323 channel driver? Regards, Jason. #--h323.conf for both [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=ulaw context=default #--dial plan of asterisk1 exten = *59,1,Wait(1) exten = *59,2,Dial(H323/[EMAIL PROTECTED]) #--dial plan of asterisk2 exten = 3500,1,Playback(hello) exten = 3500,2,Hangup() #--console messages with 'rtp debug'- -- Executing [EMAIL PROTECTED]:3] Dial(SIP/3503-0921cb88, H323/[EMAIL PROTECTED]) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Making call to [EMAIL PROTECTED]:1720 without gatekeeper. == New H.323 Connection created. -- root is calling host [EMAIL PROTECTED]:1720 -- Call token is ip$localhost/29426 -- Call reference is 29426 -- DTMF Payload is [pt=101] -- Called [EMAIL PROTECTED] Setting capabilities to 0x8 (alaw) Capabilities in preference order is (alaw) Allowed Codecs: Table: G.711-ALaw-64k 1 UserInput/hookflash 2 UserInput/RFC2833 3 UserInput/dtmf 4 Set: 0: 0: G.711-ALaw-64k 1 1: UserInput/hookflash 2 2: UserInput/RFC2833 3 UserInput/dtmf 4 -- Sending SETUP message -- Transmitting RFC2833 on payload 101 -- Started logical channel: receiving G.711-ALaw-64k -- channelsOpen = 1 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 127.0.0.1 -- remotePort: 13710 -- ExternalIpAddress: 192.168.1.116 -- ExternalPort: 29388 -- Started logical channel: sending G.711-ALaw-64k -- channelsOpen = 2 External RTP Session Starting RTP channel id 1 parameters: -- remoteIpAddress: 127.0.0.1 -- remotePort: 13710 -- ExternalIpAddress: 192.168.1.116 -- ExternalPort: 29388 - Progress Indicator: 8 -- H323/192.168.1.150-3 is making progress passing it to SIP/3503-0921cb88 -- Inbound RFC2833 on payload [pt=101] Peer capability is G.711-ALaw-64k 1 Found peer capability G.711-ALaw-64k 1, Asterisk code is 8, frame size (in ms) is 20 Peer capability is UserInput/hookflash 2 Peer capability is UserInput/RFC2833 3 Peer capability is UserInput/dtmf 4 Peer capabilities = 0x8 (alaw), ordered list is (alaw) =-= In OnConnectionEstablished for call 29426 -- Connection Established with 3500 -- H323/192.168.1.150-3 answered SIP/3503-0921cb88 -- Received Facility message... Got RTP packet from192.168.1.204:16434 (type 00, seq 014405, ts 328224084, len 000240) Sent RTP packet to 127.0.0.1:13710 (type 08, seq 008392, ts 96, len 000160) Got RTP packet from192.168.1.204:16434 (type 00, seq 014406, ts 328224324, len 000240) Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Is asterisk able to integrate with MS SQL
Yes, I have done it. I am able to connect using odbc. Now able to write to
ms sql and also retrieve in db. Now my next steps is I need to write an app
which takes a phone call, asks for the user to input a number and then
queries a MS SQL db and reads the results a row at a time back to the
caller.
anyway got example or how to go about this? I am really refresh in
programming. thanks in advance!
On 11/15/06, Wes Baehr [EMAIL PROTECTED] wrote:
Func_odbc (which is new in 1.4) was backported to 1.2. See
http://www.asterisk.org/func_odbc
While it only will return one row (there are patches to make it return
multiple rows), it's very useful for our purposes. You set up the function
in func_odbc.conf, call it with ${ODBC_FunctionName(arg1,arg2,…)} and it
executes and returns the specified data.
--
Wes Baehr
--
*From:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *On Behalf Of *Bruce Reeves
*Sent:* Wednesday, November 15, 2006 7:56 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Re: Is asterisk able to integrate with MS
SQL
I have an IVR for employees to enter certain information, like employee
number and such and then I pass that to a simple agi/php script that build
the query string and uses freetds. It took me a while to get it working and
reproduce it on several systems, but I am rather new to Linux in general.
On 11/15/06, *Tony Mountifield* [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Sharon Lim [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
Thanks, will do more research on that part. By the way, Im trying to do
IVR
where caller enter the pin the retrieve some information out of the MS
SQL.
I am wondering, what is the constraints or how to go about it. As per
said
MS SQL is about CDR. Now like i want to match and retrieve data out of
the
DB through IVR. Any guidance?
I don't think there is any direct access to MS SQL via FreeTDS from the
dialplan, but there are ODBC functions you could use. See this page:
http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc
Alternatively, implement your IVR using AGI or the ExternalIVR application
and then you can do what you like with the database.
See http://www.voip-info.org/wiki-Asterisk+AGI
and http://www.voip-info.org/wiki-Asterisk+cmd+ExternalIVR
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Bruce
Nortex Networks
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Regards,
Sharon Lim
*Good memories are to be folded neatly and tucked away into the back pocket
*
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[asterisk-users] How to kill a meet me room at midnight
Other than rebooting the server or restarting Asterisk from cron does anyone know how to kill a meetme room at midnight. Or perhaps other creative ways people deal with callers who don't hang up. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] snom subscriptions issue on WRT (2)
Nothing better, I tried some solutions, but nothing is changed. After some minutes, or after an asterisk reload , it loses all my snom subscriptions... I have an asterisk 1.2.1 on my WRT54GL , all is ok, and I use SNOM 320 as sip phones. When they boot up the subscriptions are ok, and asterisk can see their subscriptions, but when I run some of the commands above, they lose the subscriptions on asterisk. The problem is THAT: -- Reloading module 'pbx_config.so' (Text Extension Configuration) == Extension state: Watcher for hint 201 deactivated. Notify User 202 == Extension state: Watcher for hint 202 deactivated. Notify User 201 == Extension state: Watcher for hint 203 deactivated. Notify User 201 This happens when I run a reload pbx_config.so OR a RELOAD, I tried to add NAT=YES but nothing happened. The problem is that, and asterisk losts my snoms subscriptions. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why Aastra uses 48V whereas other IP Phones usemuch less, i.e. 5-12V
On Wed, Nov 22, 2006 at 11:29:01PM -0500, Michelle Dupuis wrote: 48VDC is a long time telco standard - and has become the Power over Ethernet standard. Keep in mind that 'electricity' isn't the measure - it's power. Power is not synonymous with voltage. More to the point, there is a tradeoff. For a given power, the higher the voltage, the lower the current. The lower the current, the thinner the wire you can get away with. Power over ethernet uses very thin wire, so you want high voltage and low current. Power transmission lines use very high voltage because they need (comparatively) low current through the wires. The higher the voltage, the more power you can put through the same wire. To a point. As voltage gets higher, it also gets more dangerous, and needs a bit more insulation. It's very hard to hurt somebody with 12 volts. And 48 volts, while not quite as safe, is still pretty safe. It's been chosen as a voltage that mixes the right combination of safety and power. The higher the voltage, the more heat you can generate if you have the current behind it. (If you are current limited or fuse/breaker protected you are just as safe from fire if things are calibrated right.) In the past, we often drove things with batteries, or wanted to sometimes. Getting 48v with batteries takes a lot of cells with most technologies. Phone central offices had big banks of batteries -- no problem. Today, with advanced switched-mode power supply technology, we can turn just about any voltage into any voltage. So we don't care as much about being able to run on batteries as low voltage, though it's still nice in portable tech. And of course the chips all run on very low voltages today (TTL was 5 volts and it's getting rarer) and they want to be low power.Most of the PoE phones that take 48 volts are converting it down to lower voltages to use. But 48 is a good voltage to be sending on the wires. The USA uses 120v for house current. That's enough to hurt you and can kill you if you touch it wrong, though I've touched it a few times. A lot of the world uses 220. This causes enough of a spark that they require all receptacles to have a switch on them so you don't plug things in live. On the other hand, 220 can deliver twice the power in the same current. Kettles in the 220 world are _really_ fast. Your dryer and oven run on 220 even in the 110 world, only way to get enough power. Same with electric car chargers. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to kill a meet me room at midnight
On 19:18, Thu 23 Nov 06, Eric Bishop wrote: Other than rebooting the server or restarting Asterisk from cron does anyone know how to kill a meetme room at midnight. Or perhaps other creative ways people deal with callers who don't hang up. You can use soft hangup chan -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] qualify=yes
Julian J. M. wrote: FYI, the interval at which the device is checked is 60seconds when OK, and 10s when not OK. It can be changed in channels/chan_sip.c. Look for this lines: #define DEFAULT_FREQ_OK 60 * 1000 /* How often to check for the host to be up */ #define DEFAULT_FREQ_NOTOK 10 * 1000 /* How often to check, if the host is down... */ If the device (hard or softphone) doesn't support keepalives and the nat router has a short timeout (less than 60s), even when qualify=yes, the nat mapping will timeout, thus being unable to receive calls. In this case, you can lower that 60 to a value slightly lower than the router timeout. Julian J. M. yes, but bad efect of this is, that this increase qualify check for all devices, it should be configurable in per device basis, for eg. make new option qualifycheck= in sip.conf PJ On 11/22/06, Pavel Jezek [EMAIL PROTECTED] wrote: qualify=xxx in sip means, consider peer as OK if delay reply is bellow xxx (ms) qualify checks (POKE) is every 60s (and is not configurable in sip.conf) qualify setting in iax.conf is working differently, this is how frequently to check peer (and is not possible to set some POKE delay threshlold as working qualify in sip) this is quite misleading and inconsistent and should be improved ;-) PJ Vicky wrote: I doubt that . I think qualify=500 means asterisk checks every 500 ms if the other extension is available or not . Because when qualify=( value in ms ) is set and you do a sip show peers in console asterisk whos how much latency is there between extension and asterisk . If i set qualify = no then it shows UNKNOWN . If i set qualify=10 then it doesnt mean asterisk shows extension lagged if latency is less than 10 ms ... It just checks every 10 ms for extension . I am not very sure though :) On 22/11/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Enrico Pasqualotto wrote: Enrico Pasqualotto wrote: hi all, how can I set the interval in second from retrasmit the magic packets when qualify is set to on? You have to set qualify=second instead of qualify=yes|no. This is WRONG. qualify=500 means consider this device lagged if responses take longer than 500ms I don't know if you can set the frequency of qualify packets. If you can, I assume the option would be listed in sip.conf.sample. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Diva Server, chan_capi and tone detection
This would require a change in chan-capi. To get the extended tone detection indications, additional request/parameter via CAPI must be issued. First, thanks for your reply. Do you have the CxDtmf.pdf document, from Eicon ? If I understand good, you have to enable DTMF facilities 248, 249 and 250, and then you receive DTMF code for tone detection : 0x81 for unidentified ton detected 0x80 for end of signal detected 0xC9 for human speech detected Etc... Another thing is, how do you want to get these indications for use in your dialplan? So, with DTMF code, you could handle it like for fax : redirect to extension vad or something ... Greg ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Diva Server, chan_capi and tone detection
On Thu, 23 Nov 2006, Gregory Duchatelet wrote:
This would require a change in chan-capi. To get the extended tone
detection
indications, additional request/parameter via CAPI must be issued.
First, thanks for your reply.
Do you have the CxDtmf.pdf document, from Eicon ?
Yes.
If I understand good, you have to enable DTMF facilities 248, 249 and 250,
and then you receive DTMF code for tone detection :
0x81 for unidentified ton detected
0x80 for end of signal detected
0xC9 for human speech detected
Etc...
I didn't have a closer look into the values and commands yet, but basically
that should be right.
Another thing is, how do you want to get these indications for use in
your dialplan?
So, with DTMF code, you could handle it like for fax : redirect to extension
vad or something ...
That would mean to add for each of these signals an if {} to chan-capi
source. Not very nice, but will work.
Armin
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Re: [asterisk-users] Request for working config for DISA
Hi,
Thank you for your response. As you said, I have tested. But, its not going and
simply hangup. What I have to do? Please tell me. Thank you.
Regards,
Chandra.
zero massive [EMAIL PROTECTED] wrote: Here you go:
[Custom-CLID]
exten = s,1,Answer
exten = s,2,Authenticate(12345)
exten = s,15,Playback(after-the-tone)
exten = s,16,Playback(pls-entr-num-uwish2-call)
exten = s,18,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten = s,19,Monitor(wav,${CALLFILENAME},m)
exten = s,20,DISA(no-password|from-internal|${CLIDArea})
On 11/22/06, Crazy Boy [EMAIL PROTECTED] wrote:Hi Friends,
I have configured DISA. But, its not working. When I dial to my zap channel,
its asking to enter pin number. After entering PIN number, its giving
continuous engage sound and hangup. Can anybody send me correct working
configuration for DISA? Looking forward to your response. Thank you.
Regards,
Chandra.
-
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[asterisk-users] How to change IAX default port 4569 to some other port
Hi all, All of a sudden all my IAX DIDs have gone down. I couldn't find any reason other than that the ISP is blocking port 4569. DIDs register fine from my home server, but not from office server, which is not behind any NAT. SIP registers fine. I am trying to change IAX port but it apparantly IAX works only on 4569. Changing it in iax.conf doesn't do anything. Changing it is registration string also doesn't help. How can I make IAX work on some other port? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why Aastra uses 48V whereas other IP Phones usemuch less, i.e. 5-12V
On Thu, Nov 23, 2006 at 12:40:03AM -0800, Brad Templeton wrote: [snip] The USA uses 120v for house current. That's enough to hurt you and can kill you if you touch it wrong, though I've touched it a few times. A lot of the world uses 220. This causes enough of a spark that they require all receptacles to have a switch on them so you don't plug things in live. On the other hand, 220 can deliver twice the power in the same current. Kettles in the 220 world are _really_ fast. Your dryer and oven run on 220 even in the 110 world, only way to get enough power. Same with electric car chargers. The higher the voltage, the more chance your skin will find a conductive path across the body that's dangerous. You only need 9uA across the heart and it will stop - for good. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: How to change IAX default port 4569 to some other port :Debug Message Attached
iax2 debug is giving following messages repeatedly. Tx-Frame Retry[003] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 1ms SCall: 00010 DCall: 0 [xxx.xxx.157.230:4569] USERNAME: XXX9072835 REFRESH : 60 Tx-Frame Retry[002] -- OSeqno: 002 ISeqno: 000 Type: IAX Subclass: PING Timestamp: 20001ms SCall: 6 DCall: 0 [xxx.xxx.157.230:5070] Tx-Frame Retry[003] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: LAGRQ Timestamp: 1ms SCall: 5 DCall: 0 [xxx.xxx.157.230:4569] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE110P and TDM400P
try this, pls give some feedback ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 fxsks=1-4 bchan=5-19,21-35 dchan=20 loadzone = us defaultzone=us ### On 11/22/06, Lincoln Zuljewic Silva [EMAIL PROTECTED] wrote: This is the scenarios: 1 - ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 fxsks=1-4 loadzone = us defaultzone=us ### modprobe wcte11xp ZT_CHANCONFIG failed on channel 32: No such device or address (6) FATAL: Error running install command for wcte11xp 2 - ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 fxsks=1-4 loadzone = us defaultzone=us ### modprobe wctdm ZT_CHANCONFIG failed on channel 5: No such device or address (6) FATAL: Error running install command for wctdm 3 - ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 fxsks=32-35 loadzone = us defaultzone=us ### modprobe wcte11xpok modprobe wctdmok modprobe wcfxook modprobe wct4xxpok modprobe zaptelok ### /etc/asterisk/zapata.conf [channels] context=corsidian overlapdial=yes immediate=no callprogress=yes busydetect=no switchtype=euroisdn signalling=pri_net channel = 1-15,17-31 group=2 group=1 callgroup=1 pickupgroup=1 signalling=fxs_ks channel = 32-35 ### tail -f /var/log/asterisk/messages Nov 22 15:11:43 ERROR[5524] chan_zap.c: Channel 16 is reserved for D-channel. Nov 22 15:11:43 ERROR[5524] chan_zap.c: Unable to register channel '1-15' Nov 22 15:11:43 WARNING[5524] loader.c: chan_zap.so: load_module failed, returning -1 Nov 22 15:11:43 WARNING[5524] loader.c: Loading module chan_zap.so failed! - Original Message - *From:* Henk Dick [EMAIL PROTECTED] *To:* 'Lincoln Zuljewic Silva' [EMAIL PROTECTED] ; 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com *Sent:* Wednesday, November 22, 2006 4:08 PM *Subject:* RE: [asterisk-users] TE110P and TDM400P I think that you are loading the drivers in the wrong order. You can change the order of loading are first define the E1 followed by the TDM400 Hope this helps, Henk -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Lincoln Zuljewic Silva *Sent:* woensdag 22 november 2006 20:51 *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] TE110P and TDM400P Hello all. I have here a TE110P (configured as E1) and a TDM400P (with four X100P - FXS). Both boards are recognized by the operating system as showed above: :08:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b1d9:0003 Flags: bus master, medium devsel, latency 64, IRQ 169 I/O ports at e800 [size=256] Memory at febff000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 :08:01.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device 79fe:0001 Flags: bus master, medium devsel, latency 64, IRQ 193 I/O ports at e400 [size=256] Memory at febfe000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 The problem is that I cant make the both cards to work together in the same server. Here is my /etc/zaptel.conf: ### fxsks=1-4 loadzone = us defaultzone=us span=1,1,0,ccs,hdb3,crc4 bchan=5-19,21-35 dchan=20 ### When I load the wctdm module, I get this error: ZT_CHANCONFIG failed on channel 5: No such device or address (6). Its sounds like the FXS module its tring to configure the channels 5 to 35 (E1 - ISDN Channels - TE110P card). Anybody already saw this ? Its possible to use this two cards in the same computer ? There is any separator that I can use in zaptel.conf to make the load of the modules dont mistakes itself ? Here is my versions: Debian kernel - 2.6.8 asterisk-1.2.12.1 libpri-1.2.4 zaptel-1.2.11 Thanks Lincoln ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI info
Hello, Where should I find any updated AGI informations? I am using wiki now but there are many outdated info (old pages) and might some detail changed since it written. For example I need to playback a sound file and there is a STREAM FILE command. The wiki page notice a bug but I don't know is it still exists or not because the page is nearly one year old. Luckily there is an alternative method but might I don't need that. bye, Zsolt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Request for working config for DISA
On Thu, Nov 23, 2006 at 03:05:38AM -0800, Crazy Boy wrote: Hi, Thank you for your response. As you said, I have tested. But, its not going and simply hangup. What I have to do? Please tell me. Please provide the dialplan you use as well as a trace of the CLI from when you get a call. Set verbse to at least 3. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE110P and TDM400P
On Thu, Nov 23, 2006 at 11:49:50AM +, Marco Mouta wrote: try this, pls give some feedback This one is evidently false: ### /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 It claims that the T1 span is the first one. However: fxsks=1-4 The analog span is the first one. Which is generally a bad idea, as it makes the analog card the master sync source for Zaptel (right?) bchan=5-19,21-35 dchan=20 loadzone = us defaultzone=us ### -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (OT) HylaFAX, IAXModem, Asterisk
I have all three running on the same box. I say OT because it appears asterisk is doing it's job just fine. It must be an IAXmodem or faxgetty (hylafax) problem When faxes work, they look great. I have ten IAXmodems setup with different ports and they register fine. I have ten faxgettys that startup fine. I start the IAXmodems and then faxgettys in inittab. They are setup as a roll down in the dialplan. Everything works for a while but at some point (not sure if a regular interval or random), they when a call is attempted, the channel reports back that everyone is busy. Sometimes it is just one and the next grabs the call, sometimes all of them and the call goes to congestion. Faxstat shows all faxes as running and idel and show channels in asterisk reports zero. I am losing about 40% of inbound faxes at times when the system is working and 100% when it is not. I put together the configs from many different resources found on the net since no single Howto seemed to work correctly which got me to this point. Does anyone have any clue why this happens? I have examined every log I can think of and there are no errors. Thanks, Steve Iaxmodem config: device /dev/ttyIAX0 owner uucp:uucp port4570 server 127.0.0.1 peernameiaxmodem0 secret itsasecret cidname Fax1 cidnumber 8005551212 codec slinear ttyIAX0 Config: FAXNumber: +1.800.385.7032 LongDistancePrefix: 1 InternationalPrefix:011 DialStringRules:etc/dialrules ServerTracing: 1 SessionTracing: 11 RecvFileMode: 0600 LogFileMode:0600 DeviceMode: 0600 RingsBeforeAnswer: 1 SpeakerVolume: off GettyArgs: -h %l dx_%s LocalIdentifier:NothingSetup TagLineFont:etc/lutRS18.pcf TagLineFormat: From %%l|%c|Page %%P of %%T MaxRecvPages: 100 # # # # Modem-related stuff: should reflect modem command interface # and hardware connection/cabling (e.g. flow control). # ModemType: Class1 # use this to supply a hint # # The modem is taken off-hook during initialization, and then # placed back on-hook when done to prevent glare. # ModemResetCmds: ATH1\nAT+VCID=1 # enables CallID display ModemReadyCmds: ATH0 Class1AdaptRecvCmd: AT+FAR=1 Class1TMConnectDelay: 400 # counteract quick CONNECT response Class1RMQueryCmd: !24,48,72,96 # V.17 fast-train recv doesn't work well CallIDPattern: NMBR= CallIDPattern: NAME= CallIDPattern: ANID= CallIDPattern: NDID= # Uncomment these if you really want them, but you probably don't. #CallIDPattern: DATE= #CallIDPattern: TIME= ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queuemetrics
[EMAIL PROTECTED] wrote: We are looking for a site running Queumetrics in Sydney, Australia. We have been contacted by a company in Sydney, as a few staff members of a company that are currently running Queuemetrics would like to see a fully running installation for training and decision making purposes. Their trial licence has run out and they did not test the system to the level they would have liked. Please respond to me in person if you can help. We are happy to pay for someones time on this matter. Kind regards, PaulH I use Queuemetrics in a large scale deployment, 4 million minutes a month and up to 12,000 calls a day. Works great! Historical reporting from queuemetrics is only functional for about of week due to the sheer amount of data. Direct database queries work just fine in these cases. Thanks, Steve Totaro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More than one asterisk process
I'm using 2.6 kernel on RHEL 4. Yes, all are of the same memory size. Date: Thu, 23 Nov 2006 08:20:27 +0200 From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] More than one asterisk process To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Wed, Nov 22, 2006 at 05:02:42PM -0300, Ard wrote: Hi, Can somebody in the list tell me why sometimes when I do the TOP command I see more than one asterisk process ? Sometimes it appears and desappears again... Which kernel do you use? 2.4 by any chance? If so: are all of them with the same memory size? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recordings.
On a modern server without IDE drives, you dont even need RAID to accomplish this. Problems arise at around 50-60 calls in my experience (HPDL 360, 3Ghz, Gig of RAM and RAID 1 mirroring. I run a cron job that checks files sizes and when they do not change within a specified period of time, they are considered complete and are FTPed to another server running SOX to MUX and compress the audio. Above that, checkout Orkaudio or RAMdisk. Orkaudio has my praises right now. The team over there has tweaked a recording server for us to handle about 200 simultaneous calls and all the recording is done passively through Pcap and mirrored switch ports. Thanks, Steve Vicky wrote: Hey i said that as per his requirement as an example :) . His requirement is just around 20 calls . For a moderate server i think sata raid should be fine ..Heres some result posted by someone for recording calls on ram disk . http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/118497 On 22/11/06, * Marcus Franke* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Vicky wrote: Yeh even a simple UDMA 5 enabled hard drive can handle 30 calls recording easily . Sata hard drives are even better . Hehe, UDMA sounds like EIDE drives.. nice to see they are fast enough, but I do not recommend those as server hardware. ;-) But, if John is going to buy a extra new server, he could use two drives in a mirror setup extra for recordings of these files. As it is not only the frequency of reading/writing these files but other accesses of the media like starting programs or reading/writing of logfiles that slowes down the access to the recorded audio files. Marcus -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (FreeBSD) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFFZGUUqwWWw48OFWoRAvidAJwPSpTSuY6nwxKTDKI8fZDmshmbUgCgtWAp 27akzsEDv03q5CmlGMObo50= =2jAI -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can anyone enlighten me as to what this means?
I figure the issue is probably on their side... but just want to figure out what. When you say 'users hanging up' you mean your VOIP users... or people who called in? On 11/22/06, Tristan [EMAIL PROTECTED] wrote: This happens when a call is offered to asterisk on a B-Channel that's already marked as used, I had the problem with one of my PRI provider, not hanging up calls but instead giving network congestion when users hung up... Trouble was solved at their side... Regards, Tristan Paul Hales a écrit : Are you connecting your Asterisk box to the outside world or a PABX? (I got this sort of error connecting an Asterisk box to a pabx..) PaulH On Tue, 2006-11-21 at 22:28 -0500, Matt wrote: We are doing PRIs into T4XXP cards. When I call out things are fine... however tonight sometimes on inbound calls I'd get: chan_zap.c: Duplicate setup requested on channel 0/1 already in use on span 1 in the full debug log followed by a fast busy signal on the calling parties end. Anyone know what would cause that? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RE: Snom 360 Multiple calls on hold help
Hi, try our latest beta version 6.5.2 which can be found here: http://www.snom.com/wiki/index.php/Snom360/Firmware/Beta_Versions http://www.snom.com/wiki/index.php/Snom320/Firmware/Beta_Versions http://www.snom.com/wiki/index.php/Snom300/Firmware/Beta_Versions Release Notes: http://www.snom.com/wiki/index.php/Snom360/Firmware/Release_Notes#6.5.2_beta http://www.snom.com/wiki/index.php/Snom320/Firmware/Release_Notes#6.5.2_beta http://www.snom.com/wiki/index.php/Snom300/Firmware/Release_Notes#6.5.2_beta Regards, Sven On Wednesday 22 November 2006 17:56, Ron McCarthy wrote: Yeah, doing more testing shows that the speed keys are broken, but dialing it works!!! Ugg!!! can you let me know if you get a new firmware? Im going to try and downgrade... Thanks! On 11/22/06, Alban [EMAIL PROTECTED] wrote: Yes, already. Waiting now for a new firmware... Regards, Alban Le Mercredi 22 Novembre 2006 13:50, Steve Davies a écrit: On 11/22/06, Alban [EMAIL PROTECTED] wrote: I'm having the same problem, pressing a speed dial/extension when 2 calls are on the phone connect the 2 calls together. Typing the number instead of using speed dial works. With older firmware, 6.2.1 or 6.3, it was working... But then other problem with pickup, deadlocking the phone (or slowing it down). Certainly due to the dp bug (fixed in 6.5.1). Regards, Alban. Has this been reported to snom by anyone? They are generally pretty good at fixing this type of issue and providing beta firmware. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with SER
HI, I'm not able to find some good doc or manual regarding Integration of Asterisk with SER. Bacially, I want to forward my calls from SER to asterisk. If some one already done this please guide me. thanks in advance arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE110P and TDM400P
Ok, now it works: ideiafix:~# modprobe zaptel ideiafix:~# modprobe wcte11xp ZT_CHANCONFIG failed on channel 32: No such device or address (6) FATAL: Error running install command for wcte11xp ideiafix:~# modprobe wctdm ideiafix:~# modprobe wcte11xp Order to load: zaptel, wctdm, wcte11xp Thanks a lot Henk ! - Original Message - From: Henk Dick [EMAIL PROTECTED] To: 'Lincoln Zuljewic Silva' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wednesday, November 22, 2006 5:03 PM Subject: RE: [asterisk-users] TE110P and TDM400P I would suggest the following - remove the drivers - load them manually (zaptel, wcte11xp, wctdm) Run: Zttools - should show unconfigured cards. Take: /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 fxsks=32-35 loadzone = us defaultzone=us run: ztcfg -vv See what it is saying Hope this helps ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) HylaFAX, IAXModem, Asterisk
Steve Totaro wrote: Steve, You neglet to mention: Distro Version of HylaFAX Version of iaxmodem Version of Asterisk How you're connecting to the PSTN (From previous conversations, I'm guessing PRI) I can't say that I'm not experiencing the same issue as you, 99% of our faxes are incoming. I'm running 23 iaxmodems along with HylaFAX 4.3.0.12 on a PRI and Asterisk 1.2.12.1, running on Mandriva 2006. My setting below: [iaxmodem] device /dev/ttyIAX01 port4599 refresh 60 server 127.0.0.1 record peernameiaxmodem.com01 secret 12345 cidname WhereIWork cidnumber 269xxx codec slinear [Asterisk] [iaxmodem.com01] ; Software modem COM01 type=friend host=dynamic trunk=no allowcallerid=yes disallow=all allow=slinear secret=12345 qualify=no trunk=no context=sip [HylaFAX] ModemType: Class1 # use this to supply a hint ModemSetOriginCmd: AT+VSID=%s,%d Class1AdaptRecvCmd: AT+FAR=1 Class1TMConnectDelay: 400 # counteract quick CONNECT response Class1RMQueryCmd: !24,48,72,96 # enable this to disable V.17 ModemResetCmds: AT+VCID=1 # enables CallID display -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Survey: In what ways do you use Asterisk at your house?
Earle Clubb wrote: - What service provider/technology do you use for origination/termination? - What hardware/software do you use and how does it all tie together? - What tasks do you use * to accomplish? - Any other pertinent info. Until last summer I had Asterisk doing the normal call handling my home. You know selecting which line to call out on via an SPA-3000 and SPA-3102. We do have trouble with the SPA's as the echo can be quite bad or the volume is quite low (take your pick). I'm also routing various calls to various vm-boxes and sending selected callers to the SIT. I also had an extension that interfaced to Mr. House home automation software. I could control and monitor a few things in my home. This system is no longer working due to a drive crash and the lack of backup for parts of this setup. I'm hoping to get the time towards the end of the year to put it back together. I may try to integrate the voice recognition (Sphinx) into the setup also. This was running on a 1GHz/512M/300G vanilla x86 clone. I had printer services, DNS, DHCP, file sharing, home automation, Asterisk and a few other things running. It's also my development system. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog http://home.comcast.net/~ncherry/ Backup site ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (OT) HylaFAX, IAXModem, Asterisk
Doug Lytle wrote: Steve Totaro wrote: Steve, You neglet to mention: Distro Version of HylaFAX Version of iaxmodem Version of Asterisk How you're connecting to the PSTN (From previous conversations, I'm guessing PRI) I can't say that I'm not experiencing the same issue as you, 99% of our faxes are incoming. I'm running 23 iaxmodems along with HylaFAX 4.3.0.12 on a PRI and Asterisk 1.2.12.1, running on Mandriva 2006. My setting below: [iaxmodem] device /dev/ttyIAX01 port4599 refresh 60 server 127.0.0.1 record peernameiaxmodem.com01 secret 12345 cidname WhereIWork cidnumber 269xxx codec slinear [Asterisk] [iaxmodem.com01] ; Software modem COM01 type=friend host=dynamic trunk=no allowcallerid=yes disallow=all allow=slinear secret=12345 qualify=no trunk=no context=sip [HylaFAX] ModemType: Class1 # use this to supply a hint ModemSetOriginCmd: AT+VSID=%s,%d Class1AdaptRecvCmd: AT+FAR=1 Class1TMConnectDelay: 400 # counteract quick CONNECT response Class1RMQueryCmd: !24,48,72,96 # enable this to disable V.17 ModemResetCmds: AT+VCID=1 # enables CallID display I am running on CentOS 4.4, Asterisk 1.2.10, hylafax-4.3.0-2, iaxmodem-0.1.10. Your config.tty files are much shorter than mine. I think I used the addfax script instead of copying the sample from iaxmodem. I guess it is time to upgrade a few components and try again. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error uninstalling freepbx-panel
Hi everybody, I've installed future packages (asterisk 1.2 and freepbx) from Xorcom's Repository in a debian etch, but when i want to uninstall freepbx-panel, i got this error: dialer:~# apt-get remove --purge freepbx-panel Leyendo lista de paquetes... Hecho Creando árbol de dependencias... Hecho Los siguientes paquetes se ELIMINARÃN: freepbx-panel* 0 actualizados, 0 se instalarán, 1 para eliminar y 1 no actualizados. Necesito descargar 0B de archivos. Se liberarán 65.5kB despuÃ(c)s de desempaquetar. ¿Desea continuar [S/n]? (Leyendo la base de datos ... 105470 ficheros y directorios instalados actualmente.) Desinstalando freepbx-panel ... invoke-rc.d: syntax error: missing required parameter dpkg: error al procesar freepbx-panel (--purge): el subproceso pre-removal script devolvió el código de salida de error 103 Se encontraron errores al procesar: freepbx-panel E: Sub-process /usr/bin/dpkg returned an error code (1) Any ideas how to fix this? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniería de las Telecomunicaciones PUCP Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://planeta.debianperu.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk incoming call behaviour
I am using asterisk to receive call from a DID provider . In configured everything in freepbx properly and its working . I forwarded incoming calls from did to a certain extension . Now i tried calling from another sip provider to this box , when i call from other provider to my DID number then call reaches asterisk and is sent to configured extension .. however if the extension hangs up without picking then also i am being billed at sip provider ( outgoing one ) . In simple words when people call me then they ( other people ) are billed even if configured extension isnt picked up and hangs the phone. Normally when you call a person and they hang up then you arent charged . Is this asterisk behaviour or is it freepbx dialplan the culprit here ? check your the context into which the calls are coming. if you have an answer line, there is the culprit hth ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE110P and TDM400P
On Thu, Nov 23, 2006 at 12:47:27PM -0300, Lincoln Zuljewic Silva wrote: Ok, now it works: ideiafix:~# modprobe zaptel ideiafix:~# modprobe wcte11xp ZT_CHANCONFIG failed on channel 32: No such device or address (6) FATAL: Error running install command for wcte11xp ideiafix:~# modprobe wctdm ideiafix:~# modprobe wcte11xp Order to load: zaptel, wctdm, wcte11xp No. This just goes to show that you should not run ztcfg automatically. An alternative experiment: sed -i -e 's/^install /#remmed out by Tzafrir# /' /etc/modprobe.d/zaptel wget http://svn.digium.com/svn/zaptel/team/tzafrir/zaphelper/zaptel-helper . zaptel-helper # let's get asterisk out of our way, so we could unload modules: /etc/init.d/asterisk stop # start afresh: let's make sure we have no modules loaded unload_modules # now we're ready. Feel free to add a long sleep here. # let's load the modules: modprobe wctdm modprobe wcte11xp wait_for_zap_ctl # udev may take its time generating /dev/zap/ctl ztcfg # End of experiment. However, I would still recommend that you load the wctdm driver last. This takes slight editing of zaptel.conf (and zapata.conf), or using xpp/genaptelconf . To force that load order: cat EOF /etc/modules wcte11xp wctdm EOF (or use, surprise-surprise, genzaptelconf -dM). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More than one asterisk process
On Thu, Nov 23, 2006 at 10:32:44AM -0300, Ard wrote: I'm using 2.6 kernel on RHEL 4. Yes, all are of the same memory size. That's strange. What is the output of: ps auxww | grep asterisk -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FOP is not displaying all my SIP extensions neither all E1 channels , why?
Hi, I must say that i'm not very used with customization of FOP. I've a box runing Flash Op.Panel, and i notice that the screen is full of buttons from my sip users, as well as Zapata channels. The problem is that i have more Zapata channels as well as SIP users, is there any way to get a scroll on this to display everything? do i need to resize the buttons? For sure someone now how to solve this basic question:) -- Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Terrible, horrible firewall issues in * to * setup
Lachek Butalek wrote: My mission is to get one * box to dial another * box' extensions. I have set this up previously without any issues by making a simple IAX trunk/extension pair on the two boxes and create a dial plan with a prefix like 9|XXX to select an extension on the other box. My problem is that I now have to do this with extremely restrictive firewalls thrown into the mix - firewalls I have no control over. Basically, the setup is: *1 --- FW1 --- (Internet) --- FW2 --- FW3 --- *2 I have control over firewall 1 and 3, but not 2. Using port forwarding (4569 UDP) on FW1, I have been able to make calls from *2 to *1. My problem lies with making calls the other way, as I have no way of port forwarding on FW2. My initial thought was to set up a reverse SSH tunnel from *2 to *1, which would have worked fine if SSH would tunnel UDP (latency is a different matter altogether). I found a software called Zebedee (http://www.winton.org.uk/zebedee/) which claims to do UDP tunneling, and is able to do it in reverse, but I can't for the life of me get it to work. Before I try further with Zebedee, I thought it wise to ask the * community if there is a standard solution in this particular case, or perhaps if I'm attempting the impossible. Any input is greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Try OpenVPN www.openvpn.net, *2 as client, *1 as server regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: G722?
MG == Michael Graves [EMAIL PROTECTED] writes: MG Who will benefit as long as calls must typically pass into MG existing PSTN infrstructure, and so be transcoded into G.711? It MG seems to me that only systems that are IP end-to-end stand to show MG the improvements...or am I mistunderstanding? ISDN can transport G.722. Can Digium PRI cards do G.722? /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: G722?
Benny Amorsen wrote: MG == Michael Graves [EMAIL PROTECTED] writes: MG Who will benefit as long as calls must typically pass into MG existing PSTN infrstructure, and so be transcoded into G.711? It MG seems to me that only systems that are IP end-to-end stand to show MG the improvements...or am I mistunderstanding? ISDN can transport G.722. Can Digium PRI cards do G.722? Still, I think his point is the weakest link still the PSTN hop, no matter where it will happen. If you had only VOIP end-to-end, G.722 would be good, but in any step going via PSTN (the PRI idea only solve one side of the PSTN call..unless you have the PRI in both ends of the call with G.722 ?) As people move forward into VOIP and peering, G.722 (and others) will come into play I guess, meanwhile, only for interoffice VOIP calls I guess ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium through Octasic
We're looking at using 4 or 8 port T1 cards with echo cancellation and are evaluating brands to go with. We know that Sangoma has excellent solutions especially when it comes to echo. But we still have to hear about actual performance of a Digium card using the same Octasic DSP echo canceller. Would appreciate hearing something on this. - Sponsored Link Mortgage rates near 39yr lows. $420,000 Mortgage for $1,399/mo - Calculate new house payment___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] festival problem using IAX (chan_iax2.c:2995 iax2_read)
Hi All, I'm having a problem after reinstalling the operating system. Festival works fine for SIP, but when IAX users are calling the same extension they don't hear the festival and I see the next message on console: NOTICE[3996]: chan_iax2.c:2995 iax2_read: I should never be called! I googled and couldn't find a solution, if somebody can help neobase*CLI -- Executing [EMAIL PROTECTED]:1] Goto(SIP/3001-09249a78, default|s|1) in new stack -- Goto (default,s,1) -- Executing [EMAIL PROTECTED]:1] Wait(SIP/3001-09249a78, 1) in new stack -- Executing [EMAIL PROTECTED]:2] Answer(SIP/3001-09249a78, ) in new stack -- Executing [EMAIL PROTECTED]:3] Set(SIP/3001-09249a78, TIMEOUT(digit)=5) in new stack -- Digit timeout set to 5 -- Executing [EMAIL PROTECTED]:4] Set(SIP/3001-09249a78, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing [EMAIL PROTECTED]:5] Festival(SIP/3001-09249a78, Please enter extension number) in new stack == Parsing '/etc/asterisk/festival.conf': Found -- Executing [EMAIL PROTECTED]:6] WaitExten(SIP/3001-09249a78, ) in new stack == Spawn extension (default, s, 6) exited non-zero on 'SIP/3001-09249a78' neobase*CLI neobase*CLI neobase*CLI -- Accepting UNAUTHENTICATED call from ipaddress: requested format = gsm, requested prefs = (), actual format = gsm, host prefs = (gsm), priority = mine -- Executing [EMAIL PROTECTED]:1] Wait(IAX2/ipaddress:4569-3, 1) in new stack -- Executing [EMAIL PROTECTED]:2] Answer(IAX2/ipaddress:4569-3, ) in new stack -- Executing [EMAIL PROTECTED]:3] Set(IAX2/ipaddress:4569-3, TIMEOUT(digit)=5) in new stack -- Digit timeout set to 5 -- Executing [EMAIL PROTECTED]:4] Set(IAX2/ipaddress:4569-3, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing [EMAIL PROTECTED]:5] Festival(IAX2/ipaddress:4569-3, Please enter extension number) in new stack == Parsing '/etc/asterisk/festival.conf': Found [Nov 23 18:41:41] NOTICE[3996]: chan_iax2.c:2995 iax2_read: I should never be called! -- Executing [EMAIL PROTECTED]:6] WaitExten(IAX2/ipaddress:4569-3, ) in new stack == Spawn extension (default, s, 6) exited non-zero on 'IAX2/ipaddress:4569-3' -- Hungup 'IAX2/ipaddress:4569-3' Itamar ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7970
Hello; Maybe this is a little off-topic, but I need help. I need to repair a cisco 7970, but in my country(spain) cisco is only selling, they don't repair if you're not client. Because I bought on ebay, I'm not client, so I have no chance. I tried to repair by myself, the problem is on the LCD screen, I need a replace, anyone know which part number is it (manufacturer and part number), and where I can get a replacement? Anyway, if someone knows a technical service in Spain, or Europe, where I can ask for the piece, it will help a lot. Thank you. David. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel error
http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 CP On Nov 22, 2006, at 8:40 PM, ram wrote: Hi where can i buy that Book Ram On 11/22/06, Patrick [EMAIL PROTECTED] wrote: On Wed, 2006-11-22 at 15:45 +0530, ram wrote: [snip] Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring switchtype Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring signalling Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring rxwink [snip] is this error cause any problem or just ignore this ^ Error? Where does it say error? Read the messages carefully and you will see that it says.. WARNING. If it was an error it would have said ERROR. But it didn't. Phew. Just a harmless warning. And to figure out what the warnings mean, I suggest you buy/get the Asterisk book. It's very helpful to learn about these basic things. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls from asterisk
Just use Set(CALLERID(name)) in your dialplan - that's what we do. CP On Nov 23, 2006, at 12:00 AM, Eric Bishop wrote: When we have calls that originate click-to-daial apps that use the manager interface they always originate from asterisk is there any way to change the from name? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with SER
Have a look at the OpenSER and Asterisk part of http://openser.org/dokuwiki/doku.php and http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+OpenSER Arun Kumar wrote: HI, I'm not able to find some good doc or manual regarding Integration of Asterisk with SER. Bacially, I want to forward my calls from SER to asterisk. If some one already done this please guide me. thanks in advance arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Rewriting caller ID from database?
Vincent Delporte wrote:
Hi
Most of our customers have generic names like Hospital, so I need
to
rewrite their caller ID name by looking up the number in a database
on
the Asterisk server, and rewriting the name such as Reading
Hospital
so that we know who's calling.
Any idea if this can be done with Asterisk, and how to do it?
Little C and AGI to the rescue (uses MySQL too). DB schema in the code
comments at the top.
dbc.
extensions.conf:
;; Advantech primary context (Sangoma A200D ports 12);;;
;; Primary telco number (905-xxx-)
;
[advan-primary]
exten = s,1,NoOp(Primary line - ${CALLERID}) ; write log entry
exten = s,n,agi,clid_override|${CALLERID(NUM)} ; CLID agi override
exten = s,n,Goto(cook-main-menu,s,1) ; Jump to main menu
exten = s,n,Hangup ; end/fallthrough
clid_override.c:
/* clid_override.c
* (c) Advantech Systems Integration, 2006
* Author: David B. Cook, [EMAIL PROTECTED], 905/xxx-
* Initial Delivery: Version 1.0, March 1, 2006
*
* Application to set the CLID NAME field from a local database
* when the field comes in empty from the carrier.
*
* Meant to be called from Asterisk as an AGI lookup
* Connects to MySQL database : CLID_NAME
*
* Database definition
* # Host: localhost
* # Database: asterisk
* # Table: 'CLID_NAME'
* #
* CREATE TABLE `CLID_NAME` (
* `CLID_NUM` varchar,
* `CLID_NAME` varchar,
* PRIMARY KEY (`CLID_NUM`)
* ) TYPE=InnoDB; * CLID_NAME
*
* Modification History:
* XXX 00,00 dbc - Example modification history format
*/
#include stdio.h
#include stdlib.h
#include mysql/mysql.h
#include string.h
#if !defined(MYSQL_VERSION_ID)||MYSQL_VERSION_ID32224
#define mysql_field_count mysql_num_fields
#endif
#define SELECT1_QUERY select CLID_NAME from CLID_NAME where
CLID_NUM='%s'
int main(int argc, char **argv)
{
MYSQL mysql,*sock;
MYSQL_RES *res;
MYSQL_ROW row;
char *DBhost=put hostname here;
char *DBuser=put MySQL username here;
char *DBpw=put MySQL password here;
char *DBdb=put MySQL database name here;
char qbuf[512];
int i=0;
char line[80];
/* use line buffering */
setlinebuf(stdout);
setlinebuf(stderr);
/* read and ignore AGI environment */
while (1) {
fgets(line,80,stdin);
if (strlen(line) = 1) break;
}
sprintf(qbuf,SELECT1_QUERY, argv[1]);
/* debug: show query formulation */
/* printf(SQL: %s\n, qbuf); */
/* Initialize and connect to the server */
mysql_init(mysql);
if (!(sock =
mysql_real_connect(mysql,DBhost,DBuser,DBpw,DBdb,0,NULL,0)))
{
fprintf(stderr,Couldn't connect to
engine!\n%s\n\n,mysql_error(mysql));
perror();
exit(1);
}
/* Perform query to determine if a row exists in the database for the
* CLID in question.
*/
if(mysql_query(sock,qbuf))
{
fprintf(stderr,Query 1 failed (%s)\n,mysql_error(sock));
exit(1);
}
/* No results - fatal error */
if (!(res=mysql_store_result(sock)))
{
fprintf(stderr,Couldn't get result from query failed\n,
mysql_error(sock));
exit(1);
}
if(mysql_num_rows(res)=1) {
/* CLID is PK so should only be 1 row, but I'm going to*/
/* say = just so it won't break if no PK and multiple hits. */
/* If so, will just re-set CLID again but won't break Asterisk */
while(row=mysql_fetch_row(res)) {
printf( Set VARIABLE CALLERID(name) \%s\ \n, row[0]);
/* send the output back to Asterisk */
fgets(line,80,stdin);
fputs(line,stderr);
}
}
/* Clean up memory tables/free resources */
mysql_free_result(res);
/* Terminate the database connection */
mysql_close(sock);
exit(0);
return 0; /* Keep some compilers happy */
}
___
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[asterisk-users] Cisco 7970 SIP upgrade issues
Dear Asterisk People, I am having problems putting a SIP image on a 7970. I was wondering if anyone can help? First problem is the phone is running version Load IDJar70.2-5-47-17.sbn Boot Load ID7970_64054100.bin Version5.0(0.6S) So I did read that you couldn't simply put the latest SIP image on such an old phone and a newer firmware version should be used. I got cmterm-7970_7971-sccp.7-0-2SR1 However I can't figure out how to update the firmware without a Callmanager. Can anyone enlighten me? If I do that I can then put the latest SIP image on I think Best Regards___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] When does voicemail authentication take place?
I have a rather technical question here. I'm looking at the code in app/app_voicemail.c, I'm wondering when the vmauthenticate() function is called. Aside from being called by load_module() as follows: res |= ast_register_application(app4, vmauthenticate, synopsis_vmauthenticate, descrip_vmauthenticate); I can't see any other calls to it. Can someone explain to me at what point in the program vmauthenticate() is called? Thanks so much jez Sponsored Link Rates near 39yr lows. $510,000 Loan for $1698/mo. Calcuate new payment. www.LowerMyBills.com/lre ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FREE DOWNLOAD - PRI / T1 Circuit monitoring
I have release my routines for PRI circuit monitoring. You, your client or anyone can be notified by phone, beeper, email or txtmsg that your circuit is down. If Asterisk crashes due to an oscillating circuit (as I have found it sometimes does), sendmail is usually intact and email notification and txt messages will usually get through. If the client has backup lines, and Asterisk remains up, calls can be made supplying all the information necessary to initiate a support call to the carrier, including pre dialing the support company for you or the client. These routines depend on a cron job checking the PRI status in Asterisk's database. If the database is not available, a down condition is executed. Since the database in Asterisk is used, simple modification of these routines will allow you to monitor any device and execute any type of notice you require (memory low, heavy usage...) I have not created my final web site, but rather put together a quick one which will contain more free Asterisk software and tips as time permits. http://www.siliconvp.us Sincerely, Paul Norris Owner Silicon Valley Products, Corp. attachment: winmail.dat___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] More than one asterisk process
This is the output. [EMAIL PROTECTED] ~]# ps auxw | grep asterisk root 4392 0.0 0.6 50604 13968 ? Ssl 11:02 0:00 asterisk root 5050 0.0 0.4 38416 9268 ?S11:07 0:00 asterisk root 5242 0.0 0.4 38528 9420 ?S11:09 0:00 asterisk root 5495 0.0 0.4 38448 9500 ?S11:10 0:00 asterisk root 5499 0.0 0.4 38472 9504 ?S11:10 0:00 asterisk root 5548 0.0 0.4 38404 9488 ?S11:10 0:00 asterisk root 5551 0.0 0.4 38408 9488 ?S11:10 0:00 asterisk root 5566 0.0 0.4 38360 9520 ?S11:10 0:00 asterisk root 5594 0.0 0.4 38420 9592 ?S11:10 0:00 asterisk root 5626 0.0 0.4 38512 9776 ?S11:10 0:00 asterisk root 5629 0.0 0.4 38524 9776 ?S11:10 0:00 asterisk root 5740 0.0 0.4 39528 9848 ?S11:10 0:00 asterisk root 5741 0.0 0.4 39532 9848 ?S11:10 0:00 asterisk root 5743 0.0 0.4 39540 9852 ?S11:10 0:00 asterisk root 5892 0.0 0.4 39352 9732 ?S11:10 0:00 asterisk root 5912 0.0 0.4 39332 9716 ?S11:10 0:00 asterisk root 5914 0.0 0.4 39336 9716 ?S11:10 0:00 asterisk root 7011 0.0 0.4 39828 10272 ? S11:11 0:00 asterisk From: Tzafrir Cohen [EMAIL PROTECTED] Subject: Re: [asterisk-users] More than one asterisk process To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii On Thu, Nov 23, 2006 at 10:32:44AM -0300, Ard wrote: I'm using 2.6 kernel on RHEL 4. Yes, all are of the same memory size. That's strange. What is the output of: ps auxww | grep asterisk -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Thanks for your help! Ard. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hairping calls and Originating CLI
On 22 Nov 2006, at 14:18, Adrian Marsh wrote: [Adrian Marsh] Thanks Tim, Notransfer is commented out (so I guess means = transfer). How does Asterisk know that the IN and OUT IPs are the same A*k box? (They may not be I guess). If the IPs are different, wouldn't it need to join the calls itself?? Your asterisk asks the two end points if they can/will talk to each other, if they both can, it synchronizes them, then steps out of the path. I've asked gradwell about my second point (still waiting...), but your thoughts are the same as mine. In theory it should be ok, because I have to authenticate the IAX connection with a username/password, which in turn they own and can look up if needed.. But I think theres something in UK law that says you can't be allowed to spoof the originating CLI. I don't know about a law, but the downstream interconnecting points probably make them sign contracts to that effect. Of course if you can prove to Gradwell (or whoever) that the number is yours, then it isn't spoofing - even if the call didn't really originate on that line. T. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 Error
Hello, I'm using Slackware 11.0. I've installed unixODBC from the source files. I've built and tested an odbc connection. I'm trying to install Asterisk 1.4. I can't get it to recognize the unixODBC installation. I've tried using the --with-odbc=/usr/local flag to the configure process. checking for SQLConnect in -lodbc... no configure: *** configure: *** The unixODBC installation on this system appears to be broken. configure: *** without explicitly specifying --with-odbc The above example is with the odbc flag specified to Configure. However, there's no difference in results with or without the flag. Does anyone have any idea why this isn't working? -- Richard Cook [EMAIL PROTECTED] T: 705-223-2000 ext 2010 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why Aastra uses 48V whereas other IP Phones use much less, i.e. 5-12V
On Wed, 22 Nov 2006 19:20:54 +, Steve Kennedy wrote: On Wed, Nov 22, 2006 at 06:58:13PM +0100, Huib van Wees wrote: On 11/22/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Why Aastra phones use more electricity, i.e. 48VDC whereas other phones use much less, e.g. Grandstream and Linksys both use only 5VDC. I first thought it was because of PoE, but the ones with 5VDC also run fine on PoE. What is the difference in power consumption then? 48V is also a sort of standard for telco devices if I remember it correctly... Power is nothing to do with voltage (well it is, but not alone), you need the current too i.e. V * A. Pylon electricity lines run at very high voltage (several hundred thousand volts) or the current going down the lines would heat the cables and you'd lose a lot of power. 48V is just a telco standard, and most telco equipment (that runs in racks) is 48V. Probably because 110 (or 220/240 here in EU) is enough to electrocute an engineer, and 5V/12V would require too many Amps so wiring would have to be huge to carry the current. Yes, 48v dc is a telco standard. It has to do with how they build their facilities and efficiencies in electrcal use. When your entire plant has to be on a UPS you can save much money and gain reliability by NOT having AC power supplies in every bit of gear. Thus they have standard 48v DC UPS infrastructure and everything plugs into it. This is the way to 99.999% uptime from a power perspective. It's interesting to note that outfits such as Google are now going down a similar route in planning huge new datacenters. Michael Graves ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call Transfers in SER + Asterisk architecture
Hi, I'm deploying a SER + Asterisk architecture, where SER is used as SIP registrar, and Asterisk is used for voicemail and PSTN gateway. This system is already able to make Call Transfers (Blind and Attended) internally between phones registered in SER, although, I can't make Call Transfers in some scenarios involving PSTN numbers (which need to pass through Asterisk). The problem is that when the REFER message (that carries the Refer-To number to whom the call should be transferred) gets to Asterisk, it replies with a 404 Not Found message, and the Call Transfer isn't established! Any ideas on how can I solve this problem? Thanks in advance, Ricardo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Store voicemal data in mysql DB
Hi everybody, just to confirm that I understood it right (and that the info isn't obsolete): I have to store the voicemail audio data in an external mysql DB. In http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage I read that this is only possible via ODBC and *NOT* via native mySQL (like with CDR storage). I would like to avoid using ODBC. So: Is thsi still correct? Will this change in 1.4 ? Or did I miss something? Many thanks, Norbert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 28, Issue 122
At 22:07 22/11/2006 -0700, Marco Mouta wrote: You can do it using AstDB, just load the database with callerid names and numbers and then include a lookup on database in all incoming calls, so you can override whatever you wanted:) Thanks everyone. Indeed, it seems like using the embedded BerkeleyDB-based AstDB is good enough for what I'm trying to do. However, asterisk barfs on the following script that I used to import data: #/bin/bash asterisk -rx database put cidname 1234567 'Me - cellular' asterisk -rx database put cidname 1234567 'Me - home' etc. Any idea why? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hairping calls and Originating CLI
On Thu, Nov 23, 2006 at 08:55:41PM +, Tim Panton wrote: I've asked gradwell about my second point (still waiting...), but your thoughts are the same as mine. In theory it should be ok, because I have to authenticate the IAX connection with a username/password, which in turn they own and can look up if needed.. But I think theres something in UK law that says you can't be allowed to spoof the originating CLI. I don't know about a law, but the downstream interconnecting points probably make them sign contracts to that effect. Of course if you can prove to Gradwell (or whoever) that the number is yours, then it isn't spoofing - even if the call didn't really originate on that line. You can set your CLI to whatever number is within your number range. Several providers allow you to set it to whatever you like, but they generally have an agreement (that you sign up to) that says you'll only set it to numbers you own (or are within a number range allocated to you). Just because you can set your number to something, doesn't mean you're allowed to. This became very apparent when telcos used trombing to get cheap UK termination but you had to set your origination number to your real number, and then the trombing operator would be charged the UK termination rate, not the blended rate (which is an ITU regulation). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Passing arguments to AGI script
Hi List,
Can any one please let me know how to pass arguments to the agi script from
the dialplan?
I read that it is possible to pass arguments to an AGI script here,
http://home.cogeco.ca/~camstuff/agi.html, by entering the variable followed
by a vertical bar but it doesn't seem to work for me.
I'm using a basic AGI script to query a database and then returns to
specific contexts within the dialplan in asterisk, which are set by the AGI
script depending on the results from the queries to the database.
This works fine when I use the callerid variable passed from asterisk to the
script however; I want to pass a variable to the script, which is a number
entered by the caller and stored in a varialbe within the dialplan.
Here is the part of the dialplan code calling the script,
exten = s,8,Read(options3,Test/fnnconf,1)
exten = s,9,Gotoif($[${options3} = 1]?13:10)
exten = s,10,Gotoif($[${options3} = 2]?3:11)
exten = s,11,Gotoif($[${options3} = 3]?6:12)
exten = s,12,Gotoif($[foo${options3} = foo]?t|1:i|1)
exten = s,13,agi,query.agi|${options3}
Kind Regards,
Paul
_
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[asterisk-users] Asterisk voicemail and hotel software integration
Good Evening, does anyone have information regarding integration of asterisk voicemail with an hotel management software called Fidelio made by the Micros Company. The integration can be either opensource or paid. please contact me offlist if you want. Thanks, Erick. eaperezh (at) gmail (dot) com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] FW: CISCO 7960G Asterisk
Aww, come on... not everybody has been here for ages or read through years of digests Try the voip-info WIKI: http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx Regards, Scott From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Hackensack Sent: Tuesday, November 21, 2006 10:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: CISCO 7960G Asterisk I was wondering if people have experienced issues with Cisco 7960G and Asterisk. Any feedback on people's experience deploying this phone in production environments would be appreciated. Can you at least do a search first? This same question has been asked so many times that it's been asked to death. Try something different, try Google. Sponsored Link Mortgage rates near 39yr lows. $510,000 Mortgage for $1,698/mo - Calculate new house payment http://www.lowermybills.com/lre/index.jsp?sourceid=lmb-9134-16416moid= 4119 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-users, Matt has invited you to open a Google mail account
I've been using Gmail and thought you might like to try it out. Here's an invitation to create an account. --- Matt has invited you to open a free Gmail account. To accept this invitation and register for your account, visit http://mail.google.com/mail/a-68c6563d6a-d0116c7431-a85baee643 Once you create your account, Matt will be notified with your new email address so you can stay in touch with Gmail! If you haven't already heard about Gmail, it's a new search-based webmail service that offers: - Over 2,500 megabytes (two gigabytes) of free storage - Built-in Google search that instantly finds any message you want - Automatic arrangement of messages and related replies into conversations - Powerful spam protection using innovative Google technology - No large, annoying ads--just small text ads and related pages that are relevant to the content of your messages To learn more about Gmail before registering, visit: http://mail.google.com/mail/help/benefits.html And, to see how easy it can be to switch to a new email service, check out our new switch guide: http://mail.google.com/mail/help/switch/ We're still working every day to improve Gmail, so we might ask for your comments and suggestions periodically. We hope you'll like Gmail. We do. And, it's only going to get better. Thanks, The Gmail Team (If clicking the URLs in this message does not work, copy and paste them into the address bar of your browser). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 variable list
I'd like to have a list of variables used in Asterisk 1.4, and which ones
from v1.2 were deprecated/changed.
Ex. Since switching from 1.2 to 1.4, nothing shows up when I want to display
the value of ${TIMESTAMP}.
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[asterisk-users] MWI from ITSP
How do I assign the MWI to a SIP phone on my asterisk server that is coming from an ITSP? I see the SIP message come across as having a message waiting but how does one get that to go to an extension on my box. Thanks Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Direct UA to UA RTP connection
Greetings, I have tried with all conceivable means to get my asterisk (called a in this discussion) to have two SIP user agents (called ua1 and ua2 in this discussion running SJPHONE actually) to communicate directly with one another using RTP. No matter what I do, the RTP traffic always goes between ua1 and a and a and ua2, never ua1 to ua2 directly. In my configuration a, ua1 and ua2 are all within the same network with no NAT in between. Here are the asterisk configuration settings I have: Global Nat=never (tried no also) Sip peers Nat=never (tried no also) Canreinvite=yes Once I get ua1 and ua2 to talk directly, I have another question. If a, ua1 and ua2 were all behind different NAT firewalls (ie a is in Boston, ua1 in Toronto and ua2 in San Jose), what would it take to get ua1 to RTP traffic directly to ua2. In this last scenario, ua1 and ua2 are Linksys PAP2T devices. Your expert help is greatly appreciated. Mario ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FREE DOWNLOAD - PRI / T1 Circuit monitoring
Paul wrote: I have release my routines for PRI circuit monitoring. You, your client or anyone can be notified by phone, beeper, email or txtmsg that your circuit is down. If Asterisk crashes due to an oscillating circuit (as I have found it sometimes does), sendmail is usually intact and email notification and txt messages will usually get through. If the client has backup lines, and Asterisk remains up, calls can be made supplying all the information necessary to initiate a support call to the carrier, including pre dialing the support company for you or the client. These routines depend on a cron job checking the PRI status in Asterisk’s database. If the database is not available, a down condition is executed. Since the database in Asterisk is used, simple modification of these routines will allow you to monitor any device and execute any type of notice you require (memory low, heavy usage…) I have not created my final web site, but rather put together a quick one which will contain more free Asterisk software and tips as time permits. _http://www.siliconvp.us_ Sincerely, Paul Norris Owner Silicon Valley Products, Corp. Thanks for such a useful tool and giving it away. I will try it out and give you feedback. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] aastra 480i configuration help
I had the same issue, phone was working fine but 'sip show peers' didn't show any phone registered. The reason was no sip registrar server was given in the config or in web UI. For aastra phones, you need to specify proxy and registrar servers separately. So in aastra.cfg, you need to enter the following: sip line1 registrar ip: phone.pbzinc.loc sip line1 registrar port: 5060 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Is asterisk able to integrate with MS SQL
I went with FreeTDS to accomplish this at one point and it worked great
in Dev (no call volume). It seemed to work better than ODBC since it is
speaking with M$ SQL natively rather than through an additional layer
although there is much debate about this on the net.
We were doing a bunch of local perl agi stuff too and the box started
crawling when we went live so we just went with FastAGI and a service
running on a Windows box to listen and process the FastAGI stuff, query
the M$ DB and return variables or do inserts. This approach turned out
to be lightning fast. Multiple FastAGI services can run on the same box
as long as they use different port numbers and are called by IP and port
number in your dialplan.
For your app you could use many of the standard channel variables that
are passed in FastAGI but not used (such as RDNIS in my case) to send
the data you want to interact with your DB. I have also read that in
later versions of Asterisk, you can pass other variables but I have not
really researched that since I do not need it (yet).
Also, if your app is never really going to see volume, then FreeTDS
along with local AGIs should work just fine but I would still suggest
what worked for me above. Maybe others have a better way of doing it?
I would love to hear from them.
Thanks,
Steve Totaro
Sharon Lim wrote:
Yes, I have done it. I am able to connect using odbc. Now able to
write to ms sql and also retrieve in db. Now my next steps is I need
to write an app which takes a phone call, asks for the user to input a
number and then queries a MS SQL db and reads the results a row at a
time back to the caller.
anyway got example or how to go about this? I am really refresh in
programming. thanks in advance!
On 11/15/06, * Wes Baehr* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Func_odbc (which is new in 1.4) was backported to 1.2. See
http://www.asterisk.org/func_odbc
While it only will return one row (there are patches to make it
return multiple rows), it's very useful for our purposes. You set
up the function in func_odbc.conf, call it with
${ODBC_FunctionName(arg1,arg2,…)} and it executes and returns the
specified data.
--
Wes Baehr
*From:* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]] *On Behalf Of
*Bruce Reeves
*Sent:* Wednesday, November 15, 2006 7:56 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Re: Is asterisk able to integrate
with MS SQL
I have an IVR for employees to enter certain information, like
employee number and such and then I pass that to a simple agi/php
script that build the query string and uses freetds. It took me a
while to get it working and reproduce it on several systems, but I
am rather new to Linux in general.
On 11/15/06, *Tony Mountifield* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
In article
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED],
Sharon Lim [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
Thanks, will do more research on that part. By the way, Im trying
to do IVR
where caller enter the pin the retrieve some information out of
the MS SQL.
I am wondering, what is the constraints or how to go about it. As
per said
MS SQL is about CDR. Now like i want to match and retrieve data
out of the
DB through IVR. Any guidance?
I don't think there is any direct access to MS SQL via FreeTDS
from the
dialplan, but there are ODBC functions you could use. See this page:
http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc
Alternatively, implement your IVR using AGI or the ExternalIVR
application
and then you can do what you like with the database.
See http://www.voip-info.org/wiki-Asterisk+AGI
and http://www.voip-info.org/wiki-Asterisk+cmd+ExternalIVR
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -
http://www.softins.co.uk http://www.softins.co.uk
Play: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -
http://tony.mountifield.org
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--
Bruce
Nortex Networks
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[asterisk-users] Asterisk and TDM400P ?
Hi i have buy a Digium TDM400P with 1 fxo modules TDM01B for connect my asterisk to a french analog line. In my zaptel.conf, i have: loadzone=fr defaultzone=fr fxols=3 when i load the module, i have in logs: Nov 24 06:13:40 gw kernel: Freed a Wildcard Nov 24 06:13:40 gw kernel: Zapata Telephony Interface Unloaded Nov 24 06:13:40 gw zaptel: Removing zaptel module: succeeded Nov 24 06:13:42 gw kernel: Zapata Telephony Interface Registered on major 196 Nov 24 06:13:42 gw kernel: Zaptel Version: 1.2.9.1-1mdv2007.0 Echo Canceller: KB1 Nov 24 06:13:42 gw zaptel: Loading zaptel framework: succeeded Nov 24 06:13:42 gw kernel: ACPI: PCI Interrupt :02:0a.0[A] - GSI 21 (level, low) - IRQ 20 Nov 24 06:13:43 gw kernel: Freshmaker version: 73 Nov 24 06:13:43 gw kernel: Freshmaker passed register test Nov 24 06:13:43 gw kernel: Module 0: Not installed Nov 24 06:13:43 gw kernel: Module 1: Not installed Nov 24 06:13:43 gw kernel: Module 2: Not installed Nov 24 06:13:43 gw kernel: Module 3: Installed -- AUTO FXO (FCC mode) Nov 24 06:13:43 gw kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Nov 24 06:13:44 gw kernel: Registered tone zone 2 (France) Nov 24 06:13:44 gw zaptel: Running ztcfg: succeeded and my problems are whit all sample that i have, asterisk don't restart and put me: Nov 24 06:15:17 NOTICE[2943] cdr.c: CDR simple logging enabled. Nov 24 06:15:17 WARNING[2943] chan_zap.c: Unable to specify channel 3: No such device Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to open channel 3: No such device here = 0, tmp-channel = 3, channel = 3 Nov 24 06:15:17 ERROR[2943] chan_zap.c: Unable to register channel '3' Nov 24 06:15:17 WARNING[2943] loader.c: chan_zap.so: load_module failed, returning -1 Nov 24 06:15:17 WARNING[2943] loader.c: Loading module chan_zap.so failed! for all channel (i have tested from 1 to 5) my zapata.conf: [trunkgroups] [channels] context=default signalling=fxo_ls usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes musiconhold=default channel = 3 where is my errors ? Thanks for your help ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and MISDN on a core2 Duo x64 system
Hi all i try to run misdn with asterisk on an Fedora Core 6 x64 System but after a installation of all the driver for MISDN with no errors. I get the following errors in the Full log from Asterisk logger.c: [app_exec.so]Nov 21 20:50:25 VERBOSE[21401] logger.c: [app_exec.so] = (Executes applications) Nov 21 20:50:25 VERBOSE[21401] logger.c: == Registered application 'Exec' Nov 21 20:50:25 VERBOSE[21401] logger.c: [app_readfile.so]Nov 21 20:50:25 VERBOSE[21401] logger.c: [app_readfile.so] = (Stores output of file into a variable) Nov 21 20:50:25 VERBOSE[21401] logger.c: == Registered application 'ReadFile' Nov 21 20:50:25 VERBOSE[21401] logger.c: [libisdnnet.so]Nov 21 20:50:25 WARNING[21401] loader.c: /usr/lib64/asterisk/modules/libisdnnet.so: undefined symbol: strL2State Nov 21 20:50:25 WARNING[21401] loader.c: Loading module libisdnnet.so failed! maybe I´m on the wrong way ? How can I fix this problem ? Bye MArkus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Dear Users, I am fairly new to Digium and Asterisk. I wanted to know that if I use the Digium product THREE Digium Wildcard TE412Ps (Quad E1 Card) how many calls can I handle simultaneously. I want to use the cards with the following Configurations: Intel® Xeon 3.00GHz/800MHz, 2M Processor 1GB PC2-3200 DDR2 SDRAM (2x512MB + 2X1GB) memory Integrated Dual Channel Ultra320 SCSI Adapter NC7781 Single Port PCI-X embedded NIC Hot plug drive cage - Ultra3 (6X1) High Speed IDE CD-ROM Drive 72.8GB Pluggable Ultra320 SCSI 15,000 rpm (1) Universal Hard Drive Asterisk Business Edition 3 X TE412P I have a requirement of handling 350 Calls using a single Server and please note the Server will used to transferring the call only. Other Servers will handle gateway Negotiation and Billing. This server will SIMPLY be a Gateway. Please let me know if this configuration too high or too low. If anybody has better solution please let me know that as well. Thank you, waiting eagerly for a response. Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] (no subject)
We have placed 2 X 4 port cards in a Dell 2950 and it worked well - even when it was recording 50% of the calls. PaulH On Fri, 2006-11-24 at 11:54 +0600, Imran M Yousuf wrote: Dear Users, I am fairly new to Digium and Asterisk. I wanted to know that if I use the Digium product THREE Digium Wildcard TE412Ps’ (Quad E1 Card) how many calls can I handle simultaneously. I want to use the cards with the following Configurations: Intel® Xeon™ 3.00GHz/800MHz, 2M Processor 1GB PC2-3200 DDR2 SDRAM (2x512MB + 2X1GB) memory Integrated Dual Channel Ultra320 SCSI Adapter NC7781 Single Port PCI-X embedded NIC Hot plug drive cage - Ultra3 (6X1) High Speed IDE CD-ROM Drive 72.8GB Pluggable Ultra320 SCSI 15,000 rpm (1) Universal Hard Drive Asterisk Business Edition 3 X TE412P I have a requirement of handling 350 Calls using a single Server and please note the Server will used to transferring the call only. Other Servers will handle gateway Negotiation and Billing. This server will SIMPLY be a Gateway. Please let me know if this configuration too high or too low. If anybody has better solution please let me know that as well. Thank you, waiting eagerly for a response. Imran M Yousuf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial() cmd seams unable to detect caller hangup
Dial() cmd seams unable to detect caller hangup? so if the call file land in a exten, for example: [callfile-landing] exten=1,1,dial(SIP/XXX) exten=1,n,hangup when caller after conversation and hangup, the dial cmd is unable to detect that and it will ring the caller and called party 2 times till it will give up. anyone know a fix to it? Best Regards Matt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
