Re: [asterisk-users] Ebay's SIP for Skype
I have to try Skip2PBX, integrated into my Asterisk machine, but it seem more invasive than Gizmo5 opensky. Doesn't it? Marco 2009/3/26 Grygoriy Dobrovolskyy megaho...@gmail.com skip2pbx is the best i tryed, but nasty price ;) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Early Media
I can't say it's always been like this, as I don't recall, but, Background in 1.0 behaved like this, answering the channel if it wasn't already answered and playing the sound file/s until they finished or an exten was dialed... in 1.0 the 'skip' option would cause playback to be skipped if the channel was not 'up', the 'noanswer' option would cause the channel to not be answered in 1.2 the options became 's' for skip and 'n' for noanswer though the original 'skip' and 'noanswer' options are still valid even in 1.6 That said, in this example, you'd never leave background as it would sit there playing the background_song file waiting for digits to be dialled... using the dial option would be the way... d 2009/3/27 Danny Nicholas da...@debsinc.com Is this correct for all versions, or does it start at 1.4 or 1.6? I did put a YMMV on the comment, so my answer was not to be taken as fact. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jared Smith Sent: Thursday, March 26, 2009 1:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Early Media On Wed, 2009-03-25 at 08:34 -0500, Danny Nicholas wrote: YMMV, but you might try this Exten = s,1,background(background_song) Exten = s,n,Answer() ;start billing This is not correct. Background() automatically answers the call if it hasn't been answered already. The way to accomplish the task the original poster asked is to use the m option to the Dial() application. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New CentOS 5 repo: dahdi, asterisk, freepbx RPMs
2009/3/26 John Morris aster...@zultron.com Hi, Axel. Axel Thimm wrote: How about merging in your changes/improvements/new packages with ATrpms (and automatically later into rpmrepo.org)? That way we won't have further fragmentation and a larger user base to test bits (which will be distributed in stable, testing etc repos). Of course I'd love to contribute my changes to ATrpms. Some of the small changes I made, such as adding OSLEC to the DAHDI RPMs, might be nice for ATrpms users. I'll whip up some patches against the ATrpms sources. My problem with ATrpms, though, is that the RPMs make use of many custom macros that make them unbuildable outside the ATrpms environment. I understand that might be necessary for RPMs like DAHDI that build kernel modules for several versions of several distros, where vanilla specfile code would get hairy. (I think we had this discussion a couple of years ago on the ATrpms ML.) Since I don't have to worry about multiple versions of multiple distros in my environment, I prefer to use vanilla specfile that will rebuild on anyone's CentOS 5 system. Alternatively, there's also the RPMS at http://packages.asterisk.org/centos/which seem to have a nice spread of options available, including 1.4/1.6 packages, are pretty nicely modularised and seem to be kept pretty fresh... They do however seem to have some issues that your RPMS (and Axel's) don't (e.g. why wouldn't an init file be included? and where's the changelog?)... Perhaps it would be useful to help the digium packager build some better packages... That would also help with reducing fragmentation, if there were decent quality 'official' packages available then it would save the time and effort Axel and the rpmrepo.org folks too as they could in theory base any extras on those packages rather than needing to maintain the entire set... d ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provisioning GXP 2000
On 13:45, Thu 26 Mar 09, Lutgring, Sam wrote: My preferred method is to use my own TFTP server. This makes changes to accounts/phones very fast and easy. The whole process takes me about 5 minutes to deploy an entirely new phone. 1) I modified the Grandstream template to contain my own information. This is a simple TXT document and can be edited in your favorite editor. I once counted that I am down to 8 lines in my template that need adjusting for a new user. 2) I open the above mentioned template and change the appropriate lines for the users phone and then save it to a directory utilizing a naming convention of EXTENSION-USERNAME.txt (this allows me ease of changing if ever required). 3) Then I use the Grandstream config generator to compile that into a bin file in the appropriate tftp directory. 4) Then (first time phone is ever used, not required on a redeploy) I log into the web interface on the phone and change 1 line that tells the phone where to find the config file. The phones by default allow you to use DHCP option 66 to provide the tftp server address. That will remove step 4 from your list :) 5) Reboot the phone and all done. Hope this helps. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Thursday, March 26, 2009 11:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Provisioning GXP 2000 I've done some googling and searched voip-info but I'm not able to find a good answer about how to provision the GXP 2000. Based on questions I've asked before it seems like a lot of people are using the grandstream phones so I figure provisioning can't be that hard. Is everyone using the web interface for *every* phone? Or is there a better, more automatic, way? TIA!!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ebay's SIP for Skype
2009/3/27 Marco Sambo derwid...@gmail.com I have to try Skip2PBX, integrated into my Asterisk machine, but it seem more invasive than Gizmo5 opensky. Doesn't it? Marco Skip2pbx is based on freebsd so i dont think thank you can install it on the same pc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Know who's logged in
2009/3/27 Mr. James W. Laferriere bab...@baby-dragons.com Hello Mark Miquel , On Thu, 26 Mar 2009, Mark Michelson wrote: Miguel Molina wrote: Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: # asterisk -rx show channel SIP/303-b2f1c368 -- General -- Name: SIP/303-b2f1c368 Type: SIP UniqueID: 1238094839.425549 Caller ID: 303 Caller ID Name: Ext. 303 DNID Digits: 7700 State: Up (6) Rings: 0 NativeFormats: 0x2 (gsm) WriteFormat: 0x2 (gsm) ReadFormat: 0x2 (gsm) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 111 Frames in: 6199 Frames out: 4847 Time to Hangup: 0 Elapsed Time: 3h29m16s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: XXX Extension: X Priority: XX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=X SIPURI=sip:3...@x CDR Variables: level 1: clid=Ext. 303 303 level 1: src=303 level 1: dst=XX level 1: dcontext=XXX level 1: channel=SIP/303-b2f1c368 level 1: lastapp=AgentLogin level 1: start=2009-03-26 14:13:59 level 1: answer=2009-03-26 14:13:59 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1238094839.425549 Is there an option for Agentlogin() to set a channel variable on the login channel that contains the code of the agent that successfully logged in? If not, would this be hard to accomplish by tweaking the chan_agent.c code to do that? It would be a really nice feature. I'm using asterisk 1.4.22. Thanks for any clue on this, There is a CLI command agent show which will list all agents. This output will show the agent's number, name, whether he/she is logged in, and moh class. Similarly, there is a command agent show online which will only list logged-in agents. Mark Michelson There does not seem to be a 'agent' command in 1.4.2x . asterisk-2*CLI core show version Asterisk 1.4.21.2 built by root @ asterisk-2 on a i686 running Linux on 2009-01-07 05:57:09 UTC asterisk-2*CLI agent No such command 'agent' (type 'help agent' for other possible commands) And he mentions 1.4.22 . Now unless I've misconfigured my compile of 1.4 then ... Hopefully there is a differant command ? Tia , JimL -- I would like to find a way to do it in asteris 1.2 'show agents' do not show me all agents, i have 30 agents connected to a queue and show agents show me 6 and they are offline. So is there any way to know how many agents are logged in ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX problem through intermediate asterisk box
On Thu, 26 Mar 2009, Andrew Hakman wrote:
So no one else has a problem routing IAX traffic through an
intermediate Asterisk server? Does anyone else use Asterisk in such a
configuration?
I do. Not had a problem apart from when Digium break the protocol.
1.2 - Interweb - 1.2 - Interweb - 1.2
Also now have 1.4 in the middle too.
I'm moving to SIP though because the last leg is stuck on 1.2 and carrying
the traffic is not something I want to keep on doing. (No reinvite in
IAX in 1.2)
Gordon
On Thu, Mar 26, 2009 at 2:45 AM, Andrew Hakman andrew.hak...@gmail.com
wrote:
I'm having a problem with IAX running through an intermediate asterisk
box. Perhaps a small diagram will explain the situation better:
*A --- [cloud (public internet)] --- *B [cloud
(private network)]--- *C
Asterisk server's A, B, and C, are all connected together with IAX
All asterisk servers are 1.6.0.6
Server A and B are geographically close, but connected over the public
internet.
Server B and C are geographically far, but connected over a private network.
(the latency between A and B, and B and C are roughly equal)
Each server has at least 1 phone hanging off of it, with A and C
having most of the phones (B only has a couple).
A's extensions are all 1XXX, B's are 2XXX, and C's are 3XXX
Phoning from A to B (or vice versa) works well, as does phoning from B
to C (and vice versa). Calls can be placed for an indefinite amount of
time and everything works great.
The problem arises when phoning from A through B to C (or vice versa).
For the first small amount of time (which can vary on a call to call
basis, and lasts from 0 seconds to 3 minutes or so) everything is
fine. After this, the audio in both directions gets garbled, and
starts arriving in spurts. Once this happens, it continues forever.
The audio never returns to normal no matter how long you wait.
A to B uses IAX with trunking. B to C is not using trunking
(dahdi_dummy is not working well on C for some reason - the module
loads, but no /dev/dahdi is ever created). The same behavior happens
when A to B is not using trunking either.
Usually only 1 call is being placed at a time. An interesting thing
happens when 2 testcalls are in progress at the same time though. If
there's a call from A to B, and a call from A to C is made, once the
call from A to C becomes garbled, so does the A to B call. When the A
to C call is ended, the A to B call clears up. Ending the A to B call
first does not improve the A to C call.
The dialplans are setup so each server passes all non-local extensions
to it's neighbor.
Hence, for A, the relevant part of the dialplan is
exten = _2XXX,1,Verbose(1|Extension 2xxx)
exten = _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
exten = _2XXX,n,Hangup()
exten = _3XXX,1,Verbose(1|Extension 3xxx)
exten = _3xxx,n,Dial(IAX2/asterisk_B/${EXTEN})
exten = _3xxx,n,Hangup()
For B:
exten = _1XXX,1,NoOp()
exten = _1XXX,n,Dial(IAX2/asterisk_A/${EXTEN})
exten = _1XXX,n,Hangup()
exten = _3xxx,1,NoOp()
exten = _3xxx,n,Dial(IAX2/asterisk_C/${EXTEN})
exten = _3xxx,n,Hangup()
For C:
exten = _2XXX,1,Verbose(1|Extension 2xxx)
exten = _2XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
exten = _2XXX,n,Hangup()
exten = _1XXX,1,Verbose(1|Extension 1xxx)
exten = _1XXX,n,Dial(IAX2/asterisk_B/${EXTEN})
exten = _1XXX,n,Hangup()
Is this the proper way to set such a configuration up? Is there a
better way to call from A through B to C that would work better?
Anyone else experience total audio breakup after a while with a
similar arrangement? Why does it work initially for up to about 3
minutes, then completely fall apart?
Thanks,
Andrew
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[asterisk-users] How to Integrate Neospeech with Asterisk
Hi all, I was wondering if anyone knows how to integrate the Neospeech Text to Speech engine with asterisk. I have scoured the web and haven't found anything. I think it's possible, I just don't know how to do it. If Any body tried Neospeech with Asterisk then kindly share the experience or comment. Thanks, msp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help: RED alarm on Wildcard TE122 card
Hi, I have been trying to get a Wildcard TE122 card running here the last couple of days. libpri and zaptel are all installed and configured to E1 specs. The jumper on the card is on, so configured for E1 (I'm in Norway). When running zttool, I get 'Alarms: RED' on the single card installed. Other information in zttool: Current alarms: Red Alarm Sync source: Internally clocked IRQ Misses: 1 Bipolar viol: 0 Tx/Rx levels: 1/0 Total/conf/act: 31/31/0 There are no IRQ conflicts (cat /proc/interrupt shows only wcte12xp0 on the relevant IRQ) The result is the same with or without cable inserted. The layout is: TE122-[3m cable]-[~50m cable]-[NT1 ISDN box from Telenor] When inserting the cable going into TE122 into an ISDN phone, the phone works perfectly. Any suggestions would be greatly appreciated :-) # Span 1: WCT1/0 Wildcard TE122 Card 0 (MASTER) HDB3/CCS/CRC4 RED span=1,1,1,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 # Global data loadzone= no defaultzone = no Zaptel Version: 1.4.12.1 Echo Canceller: MG2 Configuration == SPAN 1: CCS/HDB3 Build-out: 133-266 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: Clear channel (Default) (Slaves: 03) Channel 04: Clear channel (Default) (Slaves: 04) Channel 05: Clear channel (Default) (Slaves: 05) Channel 06: Clear channel (Default) (Slaves: 06) Channel 07: Clear channel (Default) (Slaves: 07) Channel 08: Clear channel (Default) (Slaves: 08) Channel 09: Clear channel (Default) (Slaves: 09) Channel 10: Clear channel (Default) (Slaves: 10) Channel 11: Clear channel (Default) (Slaves: 11) Channel 12: Clear channel (Default) (Slaves: 12) Channel 13: Clear channel (Default) (Slaves: 13) Channel 14: Clear channel (Default) (Slaves: 14) Channel 15: Clear channel (Default) (Slaves: 15) Channel 16: D-channel (Default) (Slaves: 16) Channel 17: Clear channel (Default) (Slaves: 17) Channel 18: Clear channel (Default) (Slaves: 18) Channel 19: Clear channel (Default) (Slaves: 19) Channel 20: Clear channel (Default) (Slaves: 20) Channel 21: Clear channel (Default) (Slaves: 21) Channel 22: Clear channel (Default) (Slaves: 22) Channel 23: Clear channel (Default) (Slaves: 23) Channel 24: Clear channel (Default) (Slaves: 24) Channel 25: Clear channel (Default) (Slaves: 25) Channel 26: Clear channel (Default) (Slaves: 26) Channel 27: Clear channel (Default) (Slaves: 27) Channel 28: Clear channel (Default) (Slaves: 28) Channel 29: Clear channel (Default) (Slaves: 29) Channel 30: Clear channel (Default) (Slaves: 30) Channel 31: Clear channel (Default) (Slaves: 31) 31 channels to configure. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need to find small footprint asterisk platform
Here in germany D-Link sells a device called the Horst-Box Professional wich is a ADSL modem/router with WiFi and an integrated embedded asterisk platform with 1xBRI in, 1xBRI out and 3xFXS if my mind serves me right. Size is about 180x250x50mm. Its been around for some years so maybe it is already EOL. Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: RED alarm on Wildcard TE122 card
This sounds like you have pri_net instead of pri_cpe in Zapata.conf. When inserting the cable going into TE122 into an ISDN phone, the phone works perfectly. Any suggestions would be greatly appreciated :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: RED alarm on Wildcard TE122 card
Andreas-Johann Ulvestad wrote: When inserting the cable going into TE122 into an ISDN phone, the phone works perfectly. Ummm... you have a BRI, not a PRI. I've never heard of an ISDN phone with an ISDN PRI port (E1 or T1). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: RED alarm on Wildcard TE122 card
Andreas-Johann Ulvestad schrieb: When inserting the cable going into TE122 into an ISDN phone, the phone works perfectly. That should not happen with an E1 line as your phone normally has a BRI (S0) connector with only two b-channels. Seems that your line is configured ar BRI and not PRI. Either you got the wron wire or Telenor did something terribly wrong. Hilsen ;) Christian ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf outboundproxy
John Todd wrote: Would it be so difficult to have perhaps two different proxies? One would be for any SIP messages destined for IP addresses that were not in any of the localnet= lines, and one would be for any SIP messages destined for IP addresses that were destined for IP addresses that were NOT in the localnet= lines. Of course, leaving them blank would mean that a proxy would not be used for one group or the other. This would allow creation of the concept of outside and inside at an administrative level using previously-described network definitions in sip.conf. Plus, it would dis-entangle a lot of the logic that one might otherwise have to install on the proxy to reflect certain messages back into NATted zones or otherwise complex internal structures. I don't think this is the right distinction; really, you have a list of 'known' hosts that you don't need to go through the proxy to reach, and you go through the proxy to reach the 'unknown' hosts. And, in Asterisk 1.6.x, you can already set the outboundproxy setting at the general level and on a per-peer basis. So, for all your phones/internal servers/etc., just set them to not use the proxy. In fact, this is even better when one of your 'internal' phones happens to be registered from a non-'localnet' IP address. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help: RED alarm on Wildcard TE122 card
Try to build a local loop cable first Loop pins 1-4 and 2-5 and connect to e1 port of your card. You should see the green light instead of red on card physically and ur alarm should go green too http://wiki.sangoma.com/Cablepinouts check here for cable diagram Hi, I have been trying to get a Wildcard TE122 card running here the last couple of days. libpri and zaptel are all installed and configured to E1 specs. The jumper on the card is on, so configured for E1 (I'm in Norway). When running zttool, I get 'Alarms: RED' on the single card installed. Other information in zttool: Current alarms: Red Alarm Sync source: Internally clocked IRQ Misses: 1 Bipolar viol: 0 Tx/Rx levels: 1/0 Total/conf/act: 31/31/0 There are no IRQ conflicts (cat /proc/interrupt shows only wcte12xp0 on the relevant IRQ) The result is the same with or without cable inserted. The layout is: TE122-[3m cable]-[~50m cable]-[NT1 ISDN box from Telenor] When inserting the cable going into TE122 into an ISDN phone, the phone works perfectly. Any suggestions would be greatly appreciated :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Provisioning GXP 2000
Hey, The phones we receive are all on HTTP by default and point to fm.grandstream.com by default. So I added a hosts entry to my router pointing this to my own server and my server automatically adds the mac address to the database. This way, my selects the item and says what the username and password is. David Michiel van Baak wrote: On 13:45, Thu 26 Mar 09, Lutgring, Sam wrote: My preferred method is to use my own TFTP server. This makes changes to accounts/phones very fast and easy. The whole process takes me about 5 minutes to deploy an entirely new phone. 1) I modified the Grandstream template to contain my own information. This is a simple TXT document and can be edited in your favorite editor. I once counted that I am down to 8 lines in my template that need adjusting for a new user. 2) I open the above mentioned template and change the appropriate lines for the users phone and then save it to a directory utilizing a naming convention of EXTENSION-USERNAME.txt (this allows me ease of changing if ever required). 3) Then I use the Grandstream config generator to compile that into a bin file in the appropriate tftp directory. 4) Then (first time phone is ever used, not required on a redeploy) I log into the web interface on the phone and change 1 line that tells the phone where to find the config file. The phones by default allow you to use DHCP option 66 to provide the tftp server address. That will remove step 4 from your list :) 5) Reboot the phone and all done. Hope this helps. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Ruggles Sent: Thursday, March 26, 2009 11:41 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Provisioning GXP 2000 I've done some googling and searched voip-info but I'm not able to find a good answer about how to provision the GXP 2000. Based on questions I've asked before it seems like a lot of people are using the grandstream phones so I figure provisioning can't be that hard. Is everyone using the web interface for *every* phone? Or is there a better, more automatic, way? TIA!!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk multi-cpu
Thanks. I am forced to change servers anyways, so I'm starting from scratch, which gives me the benefit of allowing me to plan things exactly as I want them. I was hoping to avoid the TC400B until the server itself was almost under strain, at which point I`d put one (or two) of those in to relieve it. But what I really wanted to know if whether I'd go with a single quad-core or two. Two isn't that much more expensive (not if it makes Asterisk process twice as much stuff) but if it doesn't add anything, I'd rather avoid this extra ~800$ per server. As for my specific needs: I am adding users/transcoded channels to this server regularly, so I do see it being not powerful enough eventually. That's why I am planning without giving any hard values: the most powerful (for the buck) the better it is. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of D Tucny Sent: Friday, March 27, 2009 0:42 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk multi-cpu From your figures, it would appear that if you double the load you will be potentially starting to see problems... FYI, not sure if it's of use to you... but... The digium tc400b is a transcoder card that can offload upto 120 channels of transcoding for g729 - ulaw... It's available as PCI only, but, if that's OK, it could be an alternative to replacing your server... G729 licenses are not needed when using that card... There have been posts by some people about having multiple CPU machines but finding that asterisk's load wasn't spread over those CPUs very well... I'm not sure if they had something special happening that caused their symptoms, but, from your dual core machine you should be able to see whether or not the load is already being spread across the 2 cores OK with your workload... d 2009/3/27 Mike l...@virtutel.ca Thanks that`s great info, and I've already subscribed to the HA mailing list. I understand call handling takes little CPU, but half my calls are transcoded from ulaw to g729 and vice versa. That seems to take my single CPU, dual-core 2.5Ghz machine up to ~35% CPU utilization. I imagine doubling what happens on my server would take me dangerously close to the upper limit of good call quality. Am I complete off? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Thursday, March 26, 2009 18:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk multi-cpu On Thu, Mar 26, 2009 at 3:06 PM, Mike l...@virtutel.ca wrote: Hi, I know somebody is going to give me the link to the wiki hardware pages, but I can't find the answer there. I'd like to know if, for an Asterisk only system (nothing else of note running on it), I get a real gain from having 2 CPUs. Does the amount of traffic/SIP registrations/codec translation possible doubles with 2 CPUs? (each quad core E5420 to be precise)? Does it increase by 50%? It is only a marginal increase, or none at all? You don't say anything about your possible kind of usage, so it's difficult to provide any specific answer to your question. In general, a few things are true: * asterisk is multi-threaded * linux kernel has nice job schedulers and i/o schedulers * if you have more ram, more things will get cached in ram * if you have more cpus / cores you can do more things at once as long as they aren't all idle waiting for some resource constraint You need to run a LOT of traffic through a server if it's just straight call handling, with a minimum of disk-bound i/o or transcoding, before you're going to max out modern hardware. So just buy the best server you want to buy, but save some money for a good warranty, or buy two servers if that's cheaper than what it would cost to be down. If you want more in-depth discussions on this you probably would prefer the asterisk-ha-clustering list: http://lists.digium.com/mailman/listinfo/asterisk-ha-clustering ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Integrate Neospeech with Asterisk
I've used NeoSpeech's Java API to build a custom TTS interface that creates sound files. I call that from Asterisk using AGI. Then I just have Asterisk play the file I created. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of msp Sent: Friday, March 27, 2009 5:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to Integrate Neospeech with Asterisk Hi all, I was wondering if anyone knows how to integrate the Neospeech Text to Speech engine with asterisk. I have scoured the web and haven't found anything. I think it's possible, I just don't know how to do it. If Any body tried Neospeech with Asterisk then kindly share the experience or comment. Thanks, msp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk multi-cpu
hi for 800 you can have a complete core 2 quad server you should have many servers and make an asterisk cluster instead of one super server. David 2009/3/27 Mike l...@virtutel.ca Thanks. I am forced to change servers anyways, so I'm starting from scratch, which gives me the benefit of allowing me to plan things exactly as I want them. I was hoping to avoid the TC400B until the server itself was almost under strain, at which point I`d put one (or two) of those in to relieve it. But what I really wanted to know if whether I'd go with a single quad-core or two. Two isn't that much more expensive (not if it makes Asterisk process twice as much stuff) but if it doesn't add anything, I'd rather avoid this extra ~800$ per server. As for my specific needs: I am adding users/transcoded channels to this server regularly, so I do see it being not powerful enough eventually. That's why I am planning without giving any hard values: the most powerful (for the buck) the better it is. Mike *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *D Tucny *Sent:* Friday, March 27, 2009 0:42 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Asterisk multi-cpu From your figures, it would appear that if you double the load you will be potentially starting to see problems... FYI, not sure if it's of use to you... but... The digium tc400b is a transcoder card that can offload upto 120 channels of transcoding for g729 - ulaw... It's available as PCI only, but, if that's OK, it could be an alternative to replacing your server... G729 licenses are not needed when using that card... There have been posts by some people about having multiple CPU machines but finding that asterisk's load wasn't spread over those CPUs very well... I'm not sure if they had something special happening that caused their symptoms, but, from your dual core machine you should be able to see whether or not the load is already being spread across the 2 cores OK with your workload... d 2009/3/27 Mike l...@virtutel.ca Thanks that`s great info, and I've already subscribed to the HA mailing list. I understand call handling takes little CPU, but half my calls are transcoded from ulaw to g729 and vice versa. That seems to take my single CPU, dual-core 2.5Ghz machine up to ~35% CPU utilization. I imagine doubling what happens on my server would take me dangerously close to the upper limit of good call quality. Am I complete off? Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Thursday, March 26, 2009 18:40 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk multi-cpu On Thu, Mar 26, 2009 at 3:06 PM, Mike l...@virtutel.ca wrote: Hi, I know somebody is going to give me the link to the wiki hardware pages, but I can't find the answer there. I'd like to know if, for an Asterisk only system (nothing else of note running on it), I get a real gain from having 2 CPUs. Does the amount of traffic/SIP registrations/codec translation possible doubles with 2 CPUs? (each quad core E5420 to be precise)? Does it increase by 50%? It is only a marginal increase, or none at all? You don't say anything about your possible kind of usage, so it's difficult to provide any specific answer to your question. In general, a few things are true: * asterisk is multi-threaded * linux kernel has nice job schedulers and i/o schedulers * if you have more ram, more things will get cached in ram * if you have more cpus / cores you can do more things at once as long as they aren't all idle waiting for some resource constraint You need to run a LOT of traffic through a server if it's just straight call handling, with a minimum of disk-bound i/o or transcoding, before you're going to max out modern hardware. So just buy the best server you want to buy, but save some money for a good warranty, or buy two servers if that's cheaper than what it would cost to be down. If you want more in-depth discussions on this you probably would prefer the asterisk-ha-clustering list: http://lists.digium.com/mailman/listinfo/asterisk-ha-clustering ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI dropping #2
This is kinda weird, but I did a fresh install of the box, upgraded from 1.4.18 to 1.4.24, replaced Zaptel with latest DAHDI. That kinda worked, but it had troubles recognizing both my TE121's, so I make a SVN checkout of DAHDI and installed that. It works fine. Not a single PRI drop in 11 hours. 2009/3/26 Harry Vangberg ha...@vangberg.name: It's 2 feet from the Nokia network terminal from the telco. 2009/3/26 Jared Smith jsm...@digium.com: On Thu, 2009-03-26 at 20:24 +0100, Harry Vangberg wrote: This sounds like what is happening, and is in order with what one of the technicians said - I was about 20 dB below their amplitude, theirs being 2048. Does this make *any* sense? How far is your Asterisk box from the demarcation point? If it's more than 133 feet (cable length), then you'll need to adjust the LBO setting on your span line in the DAHDI (or Zaptel) configuration file. -- Jared Smith Training Manager Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ATT PRI Install - What is outpulsed?
Hey All, ATT is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked her what that meant and all I got was the question repeated. Do any of you have any idea what she was referring to? Is this ANI? Outgoing Caller ID? Something else? While I've done many POTS line setups, this is my first PRI install, so I'd also welcome any make sure you do this, read this first or ATT always messes this up so... tips. Thanks -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATT PRI Install - What is outpulsed?
This is the outgoing callerid. If you have 1200 DIDs in a range, you probably only need to outpulse 4 digits (they already know the first six). If you want to be able to make your callerid anything that may or may not be one of your DIDs, you probably want 7 or 10. I pick 10 no matter what for the flexibility. -Dave -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dave Fullerton Sent: Friday, March 27, 2009 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ATT PRI Install - What is outpulsed? Hey All, ATT is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked her what that meant and all I got was the question repeated. Do any of you have any idea what she was referring to? Is this ANI? Outgoing Caller ID? Something else? While I've done many POTS line setups, this is my first PRI install, so I'd also welcome any make sure you do this, read this first or ATT always messes this up so... tips. Thanks -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATT PRI Install - What is outpulsed?
I believe she is refering to how she's going to send you your incoming calls (on your DIDs) for example: 10 digits: 972-453-2345 7 digits: 453-2345 4 digits: 2345 so you know how to expect your incoming calls and configure your extensions.conf accordingly. On Fri, Mar 27, 2009 at 10:06 AM, Dave Fullerton dfullertaster...@shorelinecontainer.com wrote: Hey All, ATT is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked her what that meant and all I got was the question repeated. Do any of you have any idea what she was referring to? Is this ANI? Outgoing Caller ID? Something else? While I've done many POTS line setups, this is my first PRI install, so I'd also welcome any make sure you do this, read this first or ATT always messes this up so... tips. Thanks -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ATT PRI Install - What is outpulsed?
Dave Fullerton wrote: Hey All, ATT is installing a PRI in a couple weeks and while I've been doing homework on PRI's for the last few weeks there's something I'm still confused about. After being asked how many digits I wanted them to send us (10) was how many digits will you outpulse to us, 4, 7 or 10? I asked Tell her 10. Most providers that I've delt with will take 7 for local or 10 for Long Distance. Some may require one or the other. You can do that easily in the dial plan. If the end user dials a 7 digit, assume to be local and tack on your local area code before the dial. Doug -- 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 http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error in ReceiveFax with T.38 -- Asterisk 1.6.0.7-rc2
On Tue, Mar 24, 2009 at 1:57 PM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
Hello,
The NoOp output was not displayed at all. I'm assuming because of the
failure in the ReceiveFax application. In fact, the verbose output
Try changing
[fax-in]
exten = 9,1,Set(INCOMING_FAXFILE=/root/santi/fax/incoming.tif)
exten = 9,n,Answer()
exten = 9,n,Wait(3)
exten = 9,n,ReceiveFax(${INCOMING_FAXFILE})
exten = 9,n,NoOp(FAXSTATUS: ${FAXSTATUS}, FAXERROR: ${FAXERROR},
FAXMODE: ${FAXMODE}, REMOTESTATIONID: ${REMOTESTATIONID}, FAXPAGES:
${FAXPAGES}, FAXBITRATE: ${FAXBITRATE}, FAXRESOLUTION:
${FAXRESOLUTION})
to
[fax-in]
exten = 9,s,1,Set(INCOMING_FAXFILE=/root/santi/fax/incoming.tif)
exten = 9,s,n,Answer()
exten = 9,s,n,Wait(3)
exten = 9,s,n,ReceiveFax(${INCOMING_FAXFILE})
exten = 9,h,1,NoOp(FAXSTATUS: ${FAXSTATUS}, FAXERROR: ${FAXERROR},
FAXMODE: ${FAXMODE}, REMOTESTATIONID: ${REMOTESTATIONID}, FAXPAGES:
${FAXPAGES}, FAXBITRATE: ${FAXBITRATE}, FAXRESOLUTION:
${FAXRESOLUTION})
exten = 9,h,HangUp
You are correct that when receivefax completes you are now in hangup context.
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Re: [asterisk-users] Know who's logged in
Mr. James W. Laferriere wrote: Hello Mark Miquel , On Thu, 26 Mar 2009, Mark Michelson wrote: Miguel Molina wrote: Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: # asterisk -rx show channel SIP/303-b2f1c368 -- General -- Name: SIP/303-b2f1c368 Type: SIP UniqueID: 1238094839.425549 Caller ID: 303 Caller ID Name: Ext. 303 DNID Digits: 7700 State: Up (6) Rings: 0 NativeFormats: 0x2 (gsm) WriteFormat: 0x2 (gsm) ReadFormat: 0x2 (gsm) WriteTranscode: No ReadTranscode: No 1st File Descriptor: 111 Frames in: 6199 Frames out: 4847 Time to Hangup: 0 Elapsed Time: 3h29m16s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: XXX Extension: X Priority: XX Call Group: 0 Pickup Group: 0 Application: AgentLogin Data: (Empty) Blocking in: ast_waitfor_nandfds Variables: AVAILSTATUS=0 AVAILORIGCHAN=SIP/303 AVAILCHAN=SIP/303-0949f890 SIPCALLID=Y2MzOTc0NmExYjVkNDNjMzhhY2I1MDMwNTk0NTJkYzQ. SIPUSERAGENT=X-Lite release 1100l stamp 47546 SIPDOMAIN=X SIPURI=sip:3...@x CDR Variables: level 1: clid=Ext. 303 303 level 1: src=303 level 1: dst=XX level 1: dcontext=XXX level 1: channel=SIP/303-b2f1c368 level 1: lastapp=AgentLogin level 1: start=2009-03-26 14:13:59 level 1: answer=2009-03-26 14:13:59 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1238094839.425549 Is there an option for Agentlogin() to set a channel variable on the login channel that contains the code of the agent that successfully logged in? If not, would this be hard to accomplish by tweaking the chan_agent.c code to do that? It would be a really nice feature. I'm using asterisk 1.4.22. Thanks for any clue on this, There is a CLI command agent show which will list all agents. This output will show the agent's number, name, whether he/she is logged in, and moh class. Similarly, there is a command agent show online which will only list logged-in agents. Mark Michelson There does not seem to be a 'agent' command in 1.4.2x . asterisk-2*CLI core show version Asterisk 1.4.21.2 built by root @ asterisk-2 on a i686 running Linux on 2009-01-07 05:57:09 UTC asterisk-2*CLI agent No such command 'agent' (type 'help agent' for other possible commands) And he mentions 1.4.22 . Now unless I've misconfigured my compile of 1.4 then ... Hopefully there is a differant command ? Tia , JimL Just typing the word agent will result in the message you see. If you press the tab key after typing the word agent you should see that one of your options for completing the command is agent show. This command is definitely in all releases of 1.4. I specifically double-checked and the command works fine for me in 1.4.22. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Diversion header
Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. ha I'm wondering if this could be used ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Diversion header
Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. Has this been replaced by something else ? Regards PS: Apologize for a previous message on this, which left my computer while I was still writing my message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Diversion header
Olivier wrote: Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. ha I'm wondering if this could be used Diversion header is an outdated draft and anyone who follows along with developments in the SIP community will tell you that other methods such as history-info are preferred over use of the diversion header. That being said, in practice, the diversion header is used by several phones. The firmware on my Polycom desk phone (IP 430) supports the sending of a Diversion header when it sends a 3XX response code. As far as Asterisk is concerned, current released versions (All 1.4 and 1.6.0) will read the Diversion header in an incoming response and use that information to fill in the rdnis of the corresponding channel's callerid structure. Once the changes from http://reviewboard.digium.com/r/201 are merged into Asterisk trunk, then Asterisk will also generate a Diversion header if you have configured Asterisk to generate redirecting information. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New CentOS 5 repo: dahdi, asterisk, freepbx RPMs
D Tucny wrote: 2009/3/26 John Morris aster...@zultron.com mailto:aster...@zultron.com Hi, Axel. Axel Thimm wrote: How about merging in your changes/improvements/new packages with ATrpms (and automatically later into rpmrepo.org http://rpmrepo.org)? That way we won't have further fragmentation and a larger user base to test bits (which will be distributed in stable, testing etc repos). Of course I'd love to contribute my changes to ATrpms. Some of the small changes I made, such as adding OSLEC to the DAHDI RPMs, might be nice for ATrpms users. I'll whip up some patches against the ATrpms sources. My problem with ATrpms, though, is that the RPMs make use of many custom macros that make them unbuildable outside the ATrpms environment. I understand that might be necessary for RPMs like DAHDI that build kernel modules for several versions of several distros, where vanilla specfile code would get hairy. (I think we had this discussion a couple of years ago on the ATrpms ML.) Since I don't have to worry about multiple versions of multiple distros in my environment, I prefer to use vanilla specfile that will rebuild on anyone's CentOS 5 system. Alternatively, there's also the RPMS at http://packages.asterisk.org/centos/ which seem to have a nice spread of options available, including 1.4/1.6 packages, are pretty nicely modularised and seem to be kept pretty fresh... They do however seem to have some issues that your RPMS (and Axel's) don't (e.g. why wouldn't an init file be included? and where's the changelog?)... Perhaps it would be useful to help the digium packager build some better packages... That would also help with reducing fragmentation, if there were decent quality 'official' packages available then it would save the time and effort Axel and the rpmrepo.org http://rpmrepo.org folks too as they could in theory base any extras on those packages rather than needing to maintain the entire set... d As the author of the RPMs at http://packages.asterisk.org/ (as well as http://packages.digium.com/), and the maintainer of the repositories, I wanted to respond to this. I would love it if some of this were to happen. I am very familiar with Axel and ATrpms - he has proven countless times that he knows what he's doing when it comes to this sort of thing. Getting help/advice from somebody like him would be extremely beneficial. As far as basing the ATrpms (or others) packages on the AsteriskNOW packages, if that is something that Axel (or others) wanted to do, I would be more than willing to help with whatever is needed. On a somewhat related, and very interesting note - I found out yesterday that the latest trixbox beta is using these RPMs (without even needing to rebuild them, in some cases). Hopefully that means I'm doing something right. D, the two issues you brought up are valid. For the Asterisk RPMs, I honestly don't know why there isn't an init script - I actually thought there was one. FreePBX is what starts Asterisk in AsteriskNOW, so it was easily overlooked. It will be there in future builds. As far as the changelog, it was one of those things that I intentionally left out for a while, and I kept meaning to do it later. Really, it's because I'm not sure what should go into an RPM changelog (I'd love to hear from anybody that has any insight into that). As always, if anybody has any ideas, suggestions, criticism, or any other type of feedback, I'd be happy to hear from you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] London DDI test request
Greetings list, I'm trying to establish if there's an issue whereby certain telcos in certain countries have not updated the London, UK numbering plan to include some parts of the 020 3 range, despite it being in operation for some two years now. To help with this, I'd be most grateful anyone outside the UK could make a test call to +44 203 3393 7389. This is a simple test number I've set up which will answer with a voicemail greeting. You'll be asked to state your country and telco. If you are unable to successfully call this number, please drop me an email to numbertest at minotaur dot it with the same information (country and telco). I'm particularly interested in callers from various providers in the USA, Australia and Switzerland, but feedback from people in all countries would be appreciated. Please do not participate if calling a UK number is going to hit your wallet in a big way when your phone bill comes round. Thanks to everyone in advance. Regards, Chris ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Integrate Neospeech with Asterisk
Can You post your solution? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* Date: Fri, 27 Mar 2009 07:55:45 -0500 From: deric.p...@nisc.coop To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to Integrate Neospeech with Asterisk I’ve used NeoSpeech’s Java API to build a custom TTS interface that creates sound files. I call that from Asterisk using AGI. Then I just have Asterisk play the file I created. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of msp Sent: Friday, March 27, 2009 5:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to Integrate Neospeech with Asterisk Hi all, I was wondering if anyone knows how to integrate the Neospeech Text to Speech engine with asterisk. I have scoured the web and haven't found anything. I think it's possible, I just don't know how to do it. If Any body tried Neospeech with Asterisk then kindly share the experience or comment. Thanks, msp _ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] London DDI test request
Quoting Chris Bagnall li...@minotaur.cc: Greetings list, I'm trying to establish if there's an issue whereby certain telcos in certain countries have not updated the London, UK numbering plan to include some parts of the 020 3 range, despite it being in operation for some two years now. To help with this, I'd be most grateful anyone outside the UK could make a test call to +44 203 3393 7389. Thins number is wrong - it has too many digits - should only be eight after the 20. (possible you put a surplus 3 in?) -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://www.tinsleyviaduct.com/phil/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 This message was sent using IMP, the Internet Messaging Program. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Weird sip problem
I've got a weird problem: I've added a new phone and sip show peers shows a status of OK (x ms) but when I dial it I get status is 'UNKNOWN' Any help on how to troubleshoot this? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 da...@safedatausa.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP for Skype Solutions: Hosted v Non-hosted
2009/3/27 Marco Sambo derwid...@gmail.com I have to try Skip2PBX, integrated into my Asterisk machine, but it seem more invasive than Gizmo5 opensky. Doesn't it? Gizmo5.com/opensky is a hosted solution SIP to Skype solution meaning there's no software to install on your system. In minutes the system can be working for your Asterisk box. This is like using Amazon's EC2 on demand computing services. Skip2PBX and other such solutions require you to install and run software - often times on an entirely different machine. And it should be noted that this machine needs to be quite powerful to perform all the transcoding involved. Sometimes this makes sense in the long run but it will have higher initial costs and setup work. It may also be more flexible. Different solutions for different needs. - -- MR Michael Robertson www.MP3tunes.com - Your Music Everywhere www.Gizmo5.com - IM/VOIP/SMS from PC and phone ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Integrate Neospeech with Asterisk
It's pretty long and involved do to a fair amount of customization we had to do. The NeoSpeech documentation includes the API and examples for using it with Java, C, .Net and COM and does a better job of explaining what you need to do than I could in a mailing list. However, if you run into specific questions, I'd be happy to do what I can in helping answer them. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Edwin Quijada Sent: Friday, March 27, 2009 11:32 AM To: Asterisk Asterisk Subject: Re: [asterisk-users] How to Integrate Neospeech with Asterisk Can You post your solution? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* Date: Fri, 27 Mar 2009 07:55:45 -0500 From: deric.p...@nisc.coop To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to Integrate Neospeech with Asterisk I've used NeoSpeech's Java API to build a custom TTS interface that creates sound files. I call that from Asterisk using AGI. Then I just have Asterisk play the file I created. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of msp Sent: Friday, March 27, 2009 5:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to Integrate Neospeech with Asterisk Hi all, I was wondering if anyone knows how to integrate the Neospeech Text to Speech engine with asterisk. I have scoured the web and haven't found anything. I think it's possible, I just don't know how to do it. If Any body tried Neospeech with Asterisk then kindly share the experience or comment. Thanks, msp Get 5 GB of storage with Windows Live Hotmail. Sign up today. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5g b_112008 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] London DDI test request
Thins number is wrong - it has too many digits - should only be eight after the 20. (possible you put a surplus 3 in?)How incredibly embarrassing. You are of course correct, try +44 20 3393 7389 :-) -Original Message- From: Phil Reynolds [mailto:phil-aster...@tinsleyviaduct.com] Sent: 27 March 2009 4:36 pm To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] London DDI test request Quoting Chris Bagnall li...@minotaur.cc: Greetings list, I'm trying to establish if there's an issue whereby certain telcos in certain countries have not updated the London, UK numbering plan to include some parts of the 020 3 range, despite it being in operation for some two years now. To help with this, I'd be most grateful anyone outside the UK could make a test call to +44 203 3393 7389. -- Phil Reynolds mail: phil-aster...@tinsleyviaduct.com Web: http://www.tinsleyviaduct.com/phil/ Waltham 66, Emley Moor 69, Droitwich 79, Windows 95 This message was sent using IMP, the Internet Messaging Program. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] London DDI test request
Am Freitag, den 27.03.2009, 16:35 + schrieb Phil Reynolds: Quoting Chris Bagnall li...@minotaur.cc: Thins number is wrong - it has too many digits - should only be eight after the 20. (possible you put a surplus 3 in?) Good guess, indeed +44 20 3393 7389 has an answering machine as announced (and can be reached from my telco, obviously). I feel some pity for the poor owner of the other number (well, minus the last digit) - he probably pulled the phone cord from the wall already. BR Anselm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Strange warning message
Can anyone give me any idea on where to start looking for this ? 1.4 svn (ish) It has appeared twice in the last hour on a system that gets numerous inbound calls to the same number TIA Julian [Mar 27 17:21:07] WARNING[3239]: ast_expr2.fl:407 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end; Input: = 2 ^ [Mar 27 17:21:07] WARNING[3239]: ast_expr2.fl:411 ast_yyerror: If you have questions, please refer to doc/channelvariables.txt in the asterisk source. __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] out of the box or do it your self?
The problem with this seems to be that when you make a distro, you want it to be many things to many people (easy to use, lots of features, support lots of hardware, you name it). When you build a medium/large call-center, you usually want to keep it lean and mean, as you need a high uptime and do not want to spend the night debugging a problem that is related to some module you do not actually need. On the other side, I'd say that we have likely hundreds of clients who build small to medium call centers with canned Asterisk distros, and they usually work reasonably well up to 50-70 seats. So the bottom line is that in most cases they are just good enough. On the other side, when they are not, the result is a blood bath - I have a few horror stories to share, but I'll keep them for Halloween :) Just my two Swiss cents, l. 2009/3/27 David fire ddf...@gmail.com hi i had installed many systems, many of them for call centers i had always installed them from scratch compiling asterisk and writing all the config from temaplates i did my self. but i saw so many out of the box solutions and i was thinking how good they are? to make one like elastix or druid (or any one) you need to know a lot of linux and asterisk so if a guy (or a group of guys) who know a lot make a distro maybe it is good enougth... David 2009/3/26 Steve Edwards asterisk@sedwards.com On Thu, 26 Mar 2009, David fire wrote: i want to ask for your opinion what is better for a call center 100 current calls and other 200 current calls make the server step by step or use a auto install cd like asterisk now, druid elastix ? and why? idontunderstandyourquestionbutithinkcaseandpuctuationmayhelp If you are asking for an opinion on whether to use an all-in-one package or build up from scratch -- it depends. If you need all the cruft on the disc, install it. It may be a prerequisite to be supported. If you don't need all the cruft or support, no. You should do a minimal server (no X) install. Meaning, de-select everything in the distro. Then, build up your installation based on your actual needs. You will end up with a more efficient and secure system that is easier and faster to maintain -- and as a bonus, you will gain an understanding into what's actually going on in your box. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Need help on how to programmatically call an extension test call state
On Thu, Mar 26, 2009 at 10:22 PM, David fire ddf...@gmail.com wrote: you can use the asterisk Manager or AMI. there is a very good java project asterisk-java but there are librarys for almost every languaje. look for Asterisk Manager and AMI www.voip-info.org is a good place to start Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] out of the box or do it your self?
This should be engraved in stone. IMHO, doing so even with a traditional telco solution would be extremely risky, if one does not have an adequate skill set and experience. Thanks l. 2009/3/26 Matt Riddell li...@venturevoip.com If you are doing an install for a call centre with 100-200 concurrent calls, you should have either done a lot of smaller installs or be working with someone who has. -- Loway - home of QueueMetrics - http://queuemetrics.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange warning message
On Fri, 2009-03-27 at 17:33 +, Julian Lyndon-Smith wrote:
Can anyone give me any idea on where to start looking for this ? 1.4
svn (ish) It has appeared twice in the last hour on a system that gets
numerous inbound calls to the same number
[Mar 27 17:21:07] WARNING[3239]: ast_expr2.fl:407 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected '=', expecting
$end; Input:
= 2
^
I typically see that when I have an expression like this:
$[${SOMEVARIABLE} = 2]
and ${SOMEVARIABLE} is empty. I'd suggest changing your expression to
look like this instead:
$[${SOMEVARIABLE} = 2]
--
Jared Smith
Training Manager
Digium, Inc.
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[asterisk-users] Six steps to better SIP security with Asterisk
In case any of you were wondering why there has been a fairly notable upswing in the attacks happening on SIP endpoints, the answer is script kiddies. In the last few months, a number of new tools have made it easy for knuckle-draggers to attack and defraud SIP endpoints, Asterisk-based systems included. There are easily-available tools that scan networks looking for SIP hosts, and then scan hosts looking for valid extensions, and then scan valid extensions looking for passwords. You can take steps, NOW, to eliminate many of these problems. I think the community is interested in coming up with an integrated Asterisk-based solution that is much wider in scope for dynamic protection (community-shared blacklists is the current thinking) but that doesn't mean you should wait for some new tool to defend your systems. You can IMMEDIATELY take fairly common-sense measures to protect your Asterisk server from the bulk of the scans and attacks that are on the increase. The methods and tools for protection already exists - just apply them, and you'll be able to sleep more soundly at night. Seven Easy Steps to Better SIP Security on Asterisk: 1) Don't accept SIP authentication requests from all IP addresses. Use the permit= and deny= lines in sip.conf to only allow a reasonable subset of IP addresess to reach each listed extension/user in your sip.conf file. Even if you accept inbound calls from anywhere (via [default]) don't let those users reach authenticated elements! 2) Set alwaysauthreject=yes in your sip.conf file. This option has been around for a while (since 1.2?) but the default is no, which allows extension information leakage. Setting this to yes will reject bad authentication requests on valid usernames with the same rejection information as with invalid usernames, denying remote attackers the ability to detect existing extensions with brute-force guessing attacks. 3) Use STRONG passwords for SIP entities. This is probably the most important step you can take. Don't just concatenate two words together and suffix it with 1 - if you've seen how sophisticated the tools are that guess passwords, you'd understand that trivial obfuscation like that is a minor hinderance to a modern CPU. Use symbols, numbers, and a mix of upper and lowercase letters at least 12 digits long. 4) Block your AMI manager ports. Use permit= and deny= lines in manager.conf to reduce inbound connections to known hosts only. Use strong passwords here, again at least 12 characters with a complex mix of symbols, numbers, and letters. 5) Allow only one or two calls at a time per SIP entity, where possible. At the worst, limiting your exposure to toll fraud is a wise thing to do. This also limits your exposure when legitimate password holders on your system lose control of their passphrase - writing it on the bottom of the SIP phone, for instance, which I've seen. 6) Make your SIP usernames different than your extensions. While it is convenient to have extension 1234 map to SIP entry 1234 which is also SIP user 1234, this is an easy target for attackers to guess SIP authentication names. Use the MAC address of the device, or some sort of combination of a common phrase + extension MD5 hash (example: from a shell prompt, try md5 -s ThePassword5000) 7) Ensure your [default] context is secure. Don't allow unauthenticated callers to reach any contexts that allow toll calls. Permit only a limited number of active calls through your default context (use the GROUP function as a counter.) Prohibit unauthenticated calls entirely (if you don't want them) by setting allowguest=no in the [general] part of sip.conf. These 7 basics will protect most people, but there are certainly other steps you can take that are more complex and reactive. Here is a fail2ban recipe ( http://www.voip-info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk ) which might allow you to ban endpoints based on volume of requests. If you'd like to see an example of the tools that you're up against, see this demo video (http://enablesecurity.com/products/enablesecurity-voippack-sipautohack-demo/ ) of an automated attack tool that does scan, guess, and crack methods via a click-and-drool interface. In summary: basic security measures will protect you against the vast majority of SIP-based brute-force attacks. Most of the SIP attackers are fools with tools - they are opportunists who see an easy way to defraud people who have not considered the costs of insecure methods. Asterisk has some methods to prevent the most obvious attacks from succeeding at the network level, but the most effective method of protection are the administrative issues of password robustness and username obscurity. JT --- John Todd email:jt...@digium.com Digium, Inc. | Asterisk Open Source Community Director 445 Jan
Re: [asterisk-users] SIP Diversion header
2009/3/27 Mark Michelson mmichel...@digium.com Olivier wrote: Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. ha I'm wondering if this could be used Diversion header is an outdated draft and anyone who follows along with developments in the SIP community will tell you that other methods such as history-info are preferred over use of the diversion header. OK, I see that RFC4244 relates to history-info. I'm adding this here for reference. That being said, in practice, the diversion header is used by several phones. The firmware on my Polycom desk phone (IP 430) supports the sending of a Diversion header when it sends a 3XX response code. As far as Asterisk is concerned, current released versions (All 1.4 and 1.6.0) will read the Diversion header in an incoming response and use that information to fill in the rdnis of the corresponding channel's callerid structure. Once the changes from http://reviewboard.digium.com/r/201 are merged into Asterisk trunk, then Asterisk will also generate a Diversion header if you have configured Asterisk to generate redirecting information. Is it planned to support in Asterisk both history-info and diversion headers ? (I can't access reviewboard at the moment soI can't check by myself). If positive, that would be interesting to know how mixed diversion/history-info hardphones are treated. Are you aware of history-info enabled (hard or soft) phone ? Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.8, and Freeplay MoH Update Released
The Asterisk development team is pleased to announce the release of Asterisk Core Sounds version 1.4.15, Extra Sounds 1.4.8, and Freeplay Music On Hold sound files. These sound files are available at http://downloads.digium.com/pub/telephony/sounds/. Future versions of Asterisk will do this automatically from the Makefile (when the sounds are enabled in menuselect). Jean-Marc Valin (from Octastic) had been experimenting with high-pass filters on the existing Asterisk sound files and found a configuration which dramatically reduced the amount of low-frequency sound on the prompts, thereby making them sound much clearer, especially when used with highly compressed codecs such as GSM and G.729. The existing sound files have been run through this filter and released back to the community. If you do not wish to upgrade your version of Asterisk, you can still install the sound prompts manually. First, download the sound files with wget for all the sound formats you need. Our example below is downloading the core sounds for the english language in the wav format: # mkdir /usr/src/asterisk-sounds # cd /usr/src/asterisk-sounds # wget http://downloads.digium.com/pub/telephony/sounds/asterisk-core-sounds-en-wav-current.tar.gz Then extract the files into your sound directory. By default in Asterisk 1.2 and 1.4, the files are located in /var/lib/asterisk/sounds/ and in Asterisk 1.6.x they are located in /var/lib/asterisk/sounds/_language_/ where _language_ should be replaced with 'en' for english, 'fr' for french, 'es' for spanish, etc. # cd /var/lib/asterisk/sounds/en/ # tar zxvf /usr/src/asterisk-prompts/asterisk-core-sounds-en-wav-current.tar.gz Then do the same for the Extra sounds. For music on hold, perform a similar process, but the music on hold sounds are located in /var/lib/asterisk/moh/. Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UPDATED: Asterisk Core Sounds 1.4.15, Extra Sounds 1.4.9, and Freeplay MoH Update Released
(Note: This announcement originally went out with an incorrect version number mentioned for the Extra sounds. It should have went out as Extra Sounds 1.4.9 and has been corrected in this announcement. Thank you for your understanding.) The Asterisk development team is pleased to announce the release of Asterisk Core Sounds version 1.4.15, Extra Sounds 1.4.9, and Freeplay Music On Hold sound files. These sound files are available at http://downloads.digium.com/pub/telephony/sounds/. Future versions of Asterisk will do this automatically from the Makefile (when the sounds are enabled in menuselect). Jean-Marc Valin (from Octastic) had been experimenting with high-pass filters on the existing Asterisk sound files and found a configuration which dramatically reduced the amount of low-frequency sound on the prompts, thereby making them sound much clearer, especially when used with highly compressed codecs such as GSM and G.729. The existing sound files have been run through this filter and released back to the community. If you do not wish to upgrade your version of Asterisk, you can still install the sound prompts manually. First, download the sound files with wget for all the sound formats you need. Our example below is downloading the core sounds for the english language in the wav format: # mkdir /usr/src/asterisk-sounds # cd /usr/src/asterisk-sounds # wget http://downloads.digium.com/pub/telephony/sounds/asterisk-core-sounds-en-wav-current.tar.gz Then extract the files into your sound directory. By default in Asterisk 1.2 and 1.4, the files are located in /var/lib/asterisk/sounds/ and in Asterisk 1.6.x they are located in /var/lib/asterisk/sounds/_language_/ where _language_ should be replaced with 'en' for english, 'fr' for french, 'es' for spanish, etc. # cd /var/lib/asterisk/sounds/en/ # tar zxvf /usr/src/asterisk-prompts/asterisk-core-sounds-en-wav-current.tar.gz Then do the same for the Extra sounds. For music on hold, perform a similar process, but the music on hold sounds are located in /var/lib/asterisk/moh/. Thank you for your continued support of Asterisk! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Diversion header
Olivier wrote: 2009/3/27 Mark Michelson mmichel...@digium.com mailto:mmichel...@digium.com Olivier wrote: Hi, Is anyone aware of SIP Diversion header ? It seems currently supported by Comverse (formely NetCentrex) softswitch and some hardphones (Thomson ST2030). An old draft (draft-levy-sip-diversion-08.txt) mentions this header. ha I'm wondering if this could be used Diversion header is an outdated draft and anyone who follows along with developments in the SIP community will tell you that other methods such as history-info are preferred over use of the diversion header. OK, I see that RFC4244 relates to history-info. I'm adding this here for reference. That being said, in practice, the diversion header is used by several phones. The firmware on my Polycom desk phone (IP 430) supports the sending of a Diversion header when it sends a 3XX response code. As far as Asterisk is concerned, current released versions (All 1.4 and 1.6.0) will read the Diversion header in an incoming response and use that information to fill in the rdnis of the corresponding channel's callerid structure. Once the changes from http://reviewboard.digium.com/r/201 are merged into Asterisk trunk, then Asterisk will also generate a Diversion header if you have configured Asterisk to generate redirecting information. Is it planned to support in Asterisk both history-info and diversion headers ? (I can't access reviewboard at the moment soI can't check by myself). If positive, that would be interesting to know how mixed diversion/history-info hardphones are treated. Are you aware of history-info enabled (hard or soft) phone ? I haven't done a lot of research with regards to history-info, so I don't know much about which phones support the feature. I don't know of any immediate plans to place history-info support into Asterisk. Of course, if the core developers were bombarded with requests to add the feature or if a community member were to write support for it into Asterisk then that would increase its consideration for inclusion. Mark Michelson ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ISDN30 Channels Locking
Hi Had an issue today where all channels connected to the telco when dialed returned WARNING[15366] chan_zap.c: Call specified, but not found? in the logs, when I removed the isdn cable and reinserted everything was fine any ideas? software Versions asterisk-1.4.21.2 zaptel-1.4.12.1 libpri-1.4.9 Thanks Robb ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE122
Does anyone know if the TE122 is recognized by any of the 1.2 zaptel drivers? It seems that 1.2.16 knows it not... :) Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE122
Jeff LaCoursiere wrote: Does anyone know if the TE122 is recognized by any of the 1.2 zaptel drivers? It seems that 1.2.16 knows it not... :) I know the TE122 is supported in Zaptel 1.2.27. Possibly supported in a few earlier versions as well. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE122
Excellent! Muchos gracias. j On Fri, 27 Mar 2009, Shaun Ruffell wrote: Jeff LaCoursiere wrote: Does anyone know if the TE122 is recognized by any of the 1.2 zaptel drivers? It seems that 1.2.16 knows it not... :) I know the TE122 is supported in Zaptel 1.2.27. Possibly supported in a few earlier versions as well. Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Six steps to better SIP security with Asterisk
On Fri, 27 Mar 2009, John Todd wrote: Seven Easy Steps to Better SIP Security on Asterisk: Six/seven -- who's counting... Thanks for this checklist. Looking forward to discussion and additions. While not specifically related to SIP, how about using autoload = no in modules.conf and only explicitly loading the modules you need -- parts left out don't get broke. I guess if you're not using SIP, leaving it out would be the ultimate in SIP security :) Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
