Re: [Asterisk-Users] asterisk gui?
Jim Van Meggelen wrote: Perhaps rather than a GUI we should be wanting an IDE (as in Integrated Development Environment, not Intelligent Drive Electronics . . . bloody overlapping acronyms . . . but I digress . . . ). Even some basic syntax highlighting would improve the readability of extensions.conf immensely. Anyone know how to make THAT work in vim? I've hacked one together for UltraEdit that works reasonably well, but that's a Windows editor. I use UltraEdit - could you share your syntax? F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] German sounds
Bastian Schern wrote: are there already some free German sounds for Asterisk? Yes, 2 sets: http://voip-info.org/wiki-Asterisk+sound+files+international F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound files - uncompressed versions available?
Holger Schurig wrote: When listening to GSM-compressed voice prompts from either G.729 or iLBC codec, the sound quality is distinctly sub-optimal due to the use of multiple transcoding. Would sox sound.gsm sound.au help a little bit? This should help with CPU usage, but not with actual sound quality - it's not possible to undo the compression artefacts :/ Thanks for the thought though :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound files - uncompressed versions available?
Hi, When listening to GSM-compressed voice prompts from either G.729 or iLBC codec, the sound quality is distinctly sub-optimal due to the use of multiple transcoding. Are the standard Asterisk sound files available in uncompressed format? - I have no problems with disk-space... PS Am aware that John Todd makes his extras available in uncompressed format: http://www.loligo.com/asterisk/sounds/AIF/ Thanks a lot, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Indications missing on Cisco FXO - * (SIP)
Fran Boon wrote: Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered SIP/10.10.2.250-9903 Looking at channel.c, I can see that this means that 'condition' is neither of 'AST_CONTROL_PROGRESS' or 'AST_CONTROL_PROCEEDING'. Presumably it's 'AST_CONTROL_RINGING', so why is this not handled? (NB Calls go through fine - all ulaw currently) Further to this, I have done more digging - it's not related to the ATA at all, but is due to the Cisco FXO port. (Calls to ATA from Firefly/IAX work fine, Calls from FXO to Firefly/IAX give this same error) I have looked at Cisco's docs they talk about using progress_ind to tune which IE is sent, but this only works for H.323, not SIP: http://cisco.com/en/US/products/sw/iosswrel/ps1839/products_command_reference_chapter09186a00800b350f.html#70 Anyone using Cisco FXO ports SIP with * getting indications? Anyone using H.323 having better luck? (If so, chan_h323 or chan_oh323?) It looks to me like a bug in * as to why this IE isn't being handled, but I could be wrong. Comments welcome :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] permission problem
Cyprien Simons wrote: Is the only way to use asterisk _not_ as root to change the permission of all the directories where asterisk need to create a file? (/var/run/, /var/log/asterisk/messages) http://voip-info.org/wiki-Asterisk+non-root F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] QoS in asterisk
[EMAIL PROTECTED] wrote: I use IAX trunking and a ping script to check times and fluctuations to my remote offices. Could you share this AGI? - seems like a useful example :) Thanks a lot, F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Indications missing on Cisco FXO - ATA-186 (SIP)
Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered SIP/10.10.2.250-9903 Looking at channel.c, I can see that this means that 'condition' is neither of 'AST_CONTROL_PROGRESS' or 'AST_CONTROL_PROCEEDING'. Presumably it's 'AST_CONTROL_RINGING', so why is this not handled? (NB Calls go through fine - all ulaw currently) Thanks a lot, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Indications missing on Cisco FXO - ATA-186 (SIP)
Rich Adamson wrote: Someone else just had that same problem in the last day or two. I don't have their response, but it had something to do with setting the Audiomode to different value to take advantage of a codec or something to that effect. Search the archives... What I saw posted recently was a problem about g.726 encoding not being supported in cvs stable then needing to turn off RFC3389 (VAD). This seems entirely different to my issue...unless I missed something? F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] collaboration with Panasonic PBX
Peter Svensson wrote: On Mon, 14 Jun 2004, Shoval Tomer wrote: Can I use a Wildcard X100P to connect an outgoing line jack (on the Panasonic) to Asterisk, so I can route calls from the PBX to Asterisk, and calls from Asterisk to the PBX? If you mean connecting the X100P to an analog extension line then that will work both for incoming and outgoing. Note that the KX-TD1232 analog lines do not provide caller id, at least ours do not. That's a shame- what protocol do they use? DTMF? http://voip-info.org/wiki-Asterisk+bounty+non-Bellcore-CLID Another option could be to connect Asterisk using an internal isdn extension. We have a few isdn modems hanging off our pbx that way and they get callerid etc so Asterisk should be able to as well. We interface Asterisk to our pbx using a pri line instead so I have not tried using a bri line myself. Do you use the TD-1232's 'T1' interface, then? - with what PRI card? Digium or Cisco or what? - does it support Q.931? Their webpage is vague as to exactly what they mean by 'T1' http://catalog2.panasonic.com/webapp/wcs/stores/servlet/ModelDetail?displayTab=FstoreId=11251catalogId=11005itemId=62983catGroupId=2modelNo=KX-TD1232surfModel=KX-TD1232ignoreRedirect=1 Thanks for any extra information - I need to interface * with one of these in 2 locations. F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mine strangest asterisk problem ever ....
Alessio Focardi wrote: BF You can try doing different things with it, but I know that I am currently BF set to level 3 rather than 5 as default with RedHat. I checked hdparm googling around, what parameter have you set to 3 instead of 5 ? I'm pretty sure this is a confusion. I think this must refer to runlevel 5 - 3 i.e. not having X running... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help, Ideas and Ready for use Solutions
Bisker, Scott (7805) wrote: If there is already an existing phone system in place, you could easily migrate to an asterisk based solution if your internal phones are analog. The big question for you is not number of phone lines, but peak utilization. Here's what I have. Max concurrent calls 15-20 (30-40 active channels) How do you measure the peak concurrent calls/active channels? Do you have a script for getting this out of the CDRs? Thanks, F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DNS SRV records
On Wed, 2004-06-02 at 13:40, Andrew Thompson wrote: My DNS gui(Cpanel/WHM) only allows the following options for entry type: A6 CNAME MX NS PTR TXT WRK Does anyone know if any of these options are acceptable substitutes for an SRV record, or do I need to put in a ticket to have a SRV record specifically created for me? Sorry, really needs to be SRV :/ F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem compiling ZAPTEL on Linux 2.6.6
On Wed, 2004-06-02 at 14:56, Miroslav Nachev wrote: I have Debian Linux with kernel 2.6.6. The all packages compiled except ZAPTEL where I have the following error: voipgw:/usr/src/zaptel# make make linux26 F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk process respawn
On Wed, 2004-06-02 at 17:33, Steven Critchfield wrote: On Wed, 2004-06-02 at 10:01, Terry Goodwin wrote: Anyone know how to place asterisk in initab so that it is loaded at boot and will respawn if the process goes down? Don't put it in initab, use the startup script already provided with asterisk. safe_asterisk respawns itself already optionally emails you to let you know that it needed to. v.nice :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] @mydomain.com
On Tue, 2004-06-01 at 10:11, Simon Chappell wrote: I noticed that alot of people are displaying sip:[EMAIL PROTECTED] Can I achieve this with asterisk or do i need something else? Sure :) I have a domain and spare IP's so the dns is not a problem. Just create SRV records in your DNS to point SIP at your Asterisk server's public IP address F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] @mydomain.com
On Tue, 2004-06-01 at 14:03, Simon Chappell wrote: I assume thta i need to open port 5060 also? Yes also the appropriate RTP ports (unless your Firewall/NAT is SIP-aware can open RTP ports based on SIP messages...) F Fran Boon wrote: On Tue, 2004-06-01 at 10:11, Simon Chappell wrote: I noticed that alot of people are displaying sip:[EMAIL PROTECTED] Can I achieve this with asterisk or do i need something else? Sure :) I have a domain and spare IP's so the dns is not a problem. Just create SRV records in your DNS to point SIP at your Asterisk server's public IP address F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] @mydomain.com
On Tue, 2004-06-01 at 16:00, Simon Chappell wrote: would the number be the extension? or the number i have at fwd.. ie sip:[EMAIL PROTECTED] or sip:[EMAIL PROTECTED] Your extension ;) or nicer if you can set up an alias. e.g. exten = schappell,1,Goto(lan,2000) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GnoPhone
On Tue, 2004-06-01 at 21:25, Stuart Grimshaw wrote: I'm currently trying to find a soft phone that will work properly of my Gentoo based laptop. Try IAXComm: http://iaxclient.sf.net/ F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] extracting country code from a number
On Sun, 2004-05-30 at 10:00, usedcanon wrote: Obviously if there is something there I am not entering the right search criteria. Further help will be appreciated. May 10th asterisk-dev archives. Post by Rob Gagnon: Algorithm to parse country code F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distribution of Linux
On Sun, 2004-05-30 at 17:47, two wrote: I'm using Asterisk by Red Hat Linux 9 now. Can Asterisk also use Fedora Linux 1 and Fedora Linux 2? Yes Yes Getting Zaptel to compile with FC2 is a bit of a hassle, but can be done http://voip-info.org/wiki-Asterisk+linux+distributions F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Caller ID with BT CD50
On Fri, 2004-05-28 at 10:27, Kevin Walsh wrote: Chris Stenton [EMAIL PROTECTED] wrote: Could you add this to http://bugs.digium.com/bug_view_page.php?bug_id=0001719 I thought that, if it was confirmed as working for people other than just me, Tony Hoyle might want to add it to his original patch. People could then apply his single master patch, rather than a handful of patches to patches. Yes, but it can start with you adding as-is to the bugtracker confirming that you've sent in a disclaimer. Then it can safely be merged into 1 patch. F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Database
On Fri, 2004-05-28 at 16:10, Ed Devine wrote: I'd like to be able to add additional fields to the the Asterisk database. I'm using Mysql for most of my data lookup and manipulation, and it seems to work pretty well. In keeping with what I know how to do, it would be very handy to be able to insert say a call forward number into a customer record. That way, I could automatically route calls to extensions to a forwarded number. Any suggestions on how this can be done? http://voip-info.org/wiki-Asterisk+configuration+from+database I use method 2 to #include sections into my user configs dialplan I complement this with app_dbodbc to do lookups in my dial macro: http://voip-info.org/wiki-Asterisk+app_dbodbc Dial macro then ensures that users' original voicemail is accessed if the fwd'd extension is busy. F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CTI (Computer-Telephony Integration) with Asterisk ?
Florent Guiliani wrote: http://www.automated.it/guidetoasterisk.html Error 404 :-( http://www.automated.it/guidetoasterisk.htm ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_odbc with mysql on a remote server
Adam Goryachev wrote: So, the problem I am having is that the mysql odbc driver seems to want to use a local socket, but I am not running mysql locally on the asterisk machine. I want it to connect to a remote host. This is an ODBC issue, not an Asterisk issue. Check odbc.ini F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 11 instead of Star
Greg Blakely wrote: Well, I did a search and replace on chan_zap.c, and got most of it converted to 11XX instead of *XX, but the call pickup code still eludes me. Is it set somewhere else? res/res_parking.c F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ZAPTEL not loading on FC2
Jorge Verastegui wrote: I have successfully compiled the last cvs zaptel drives in FC2 box and then load wcfxs module, but Kernel Freezes with zttool http://bugs.digium.com/bug_view_page.php?bug_id=0001704 F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme Options (new one)
Ben Merrills wrote: Seems like it would be a simple modification? Where would I post a feature request like this? :-) bugs.digium.com Ensure summary starts with [request] F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip Registration Problem
I removed the qualify lines and sip reload [ed]. The extension still showed up as UNREACHABLE instead of UNMONITORED. I had to do a full restart to get it to stop sending the OPTIONS messages. What did I do wrong here? How can I make a change to qualify without restarting? If a peer is registred at reload/sip reload, it will not change. You have to unload the sip module and reload it or restart asterisk to change the configuration of a registred, i.e. active, peer. /O Brett Nemeroff wrote: How will this effect a live system? No new calls? Or will it terminate exisiting calls? Unloading SIP module will terminate all SIP calls Restarting Asterisk will terminate all calls :( ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SMDI support in Asterisk ?
W. Kevin Hunt wrote: I'll add $1k to that bounty, and will put another bounty out for $3k for ss7 integration w/ full isup / imt support... John Bittner wrote: I am also looking for the SMDI support. I am willing to put up a bounty of 2K to get this writen. Anyone interested please email me off list. ok, I've added these to the Wiki: http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SMDI http://voip-info.org/tiki-index.php?page=Asterisk+bounty+SS7 Anyone who has more info on what needs doing should add info there, also any further contributions... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Admission Control
Rana Dutt wrote: Let's say you have a 256 Kbps Internet connection and you're using it for voice calls. With mu-law (G.711), each call uses about 80 kbps, so you really can't have more than 3 calls active at one time. Does Asterisk support any kind of Call Admission Control where it would prevent you from originating a call if it would exceed your Internet bandwidth? For example, in this case, ideally, we would want Asterisk to present busy tone when the fourth simultaneous call is attempted. Not quite :/ The closest thing is app_groupcount (available in CVS HEAD) This allows you to restrict outbound calls, but incoming calls are harder to restrict. There is also an incominglimit outgoinglimit for sip.conf http://voip-info.org/wiki-Asterisk+sip+incominglimit It says there that 'outgoinglimit' is currently disabled in the source code, but I don't see this when I look at the code. It does say that app_groupcount is to replace this completely. I would really like to see a 'total calls' limit for my remote servers (so bandwidth-limited that this will often be just 1!) Currently I can only set a maximum of 1 IAX call in each direction (since I control both ends of the trunk). I can't get it down to 1 in total... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] extensions/sip from database?
Manuel Wenger wrote: We are planning to deploy a pretty large asterisk server with many SIP extensions (might be up to 1 in the future), and I have a few questions: 1) is this possible, or are we running into some kind of limitation in the software that I wasn't aware of and that I didn't find by browsing through the archives and through Wiki? No, we don't need any G729-G711 transformations, it would only be acting as a SIP proxy (even if asterisk isn't a proxy). /Should/ be psosible with canreinvite=yes no use of T,t in the dial commands, so that Asterisk can stay out of the media path except when absolutely necessary. 2) is there a way to store extensions.conf and/or sip.conf in some kind of database, maybe MySQL? This would make life easier if someone wanted to change his SIP password. Or how would you otherwise solve this problem? http://voip-info.org/wiki-Asterisk+configuration+from+database Option 1 is being enhanced through the development of ast_data. I currently use Option 2 3) is there a quick way of reloading only a part of sip.conf/extensions.conf, for example if only a user password changed, or an extension's behaviour (eg. routing to voicemail instead of a SIP user)? sip reload extensions reload That's as granular as it gets. Should be harmless to keep doing this, though. Maybe I'm looking at the wrong software here and SER would be better for what I want to do... I know asterisk is supposed to be a PBX replacement, but the functions and flexibility it has really tells me stick with asterisk. Or am I way off with these assumptions? Possibly - depends whether you're after a SIP proxy or a PBX ;) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dynamic SIP.CONF
Darren Nay wrote:
We are looking to expand our usage of Asterisk and I am trying to make as
much of the configuration dynamic as I possibly can. The only part that I'm
having problems with is sip.conf. I can get asterisk to register each
extension with our local SER SIP proxy dynamically by using the
sipfriends table in the database, but I'm having trouble with the message
waiting indicators (ie. SIP NOTIFY packets when a new voicemail is waiting).
-SNIP-
Is there a way to make this dynamic so that I don't have to add this into
sip.conf -every- single time that I add a new extension?
Only by extending the functionality of sip friends to include this extra
field...
I wouldn't bother doing this as ast_data (formally res_data) is being
developed to replace sip/iax friends.
If you want to take a sneak preview at this then see:
http://svn.asteriskdocs.org/res_data/ast_data/
I tried the following, but it didn't work ..
[default]
type=peer
host=dynamic
dtmfmode=inband
username=${EXTEN}
Mailbox=${EXTEN}
Am I on the right track, or way off base? :-)
Way off base ;)
That kind of syntax only works in extensions.conf
F
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Re: [Asterisk-Users] Dynamic SIP.CONF
Brian Cuthie wrote: So I've been kind of struggling with the notion of making my Asterisk implementations dynamic, too. While I'd like to make everything directly database driven, I'm not sure Asterisk is quite there yet. I've been thinking of writing something that creates appropriate configuration files from the database on a periodic basis, and then does an Asterisk reload. This would introduce a small delay into configuration changes, but it does have other benefits such as decoupling the design of the database from Asterisk. Any thoughts? This is exactly what I do - works very well so far :) I guess that it will reach scalability limits at some stage...but so far, so good... I write out: users-sip.conf users-iax.conf users-voicemail.conf mapping.conf(username- extension) These are #included into the main files. I restart Asterisk via the manager port, since 'asterisk -r -x reload' doesn't return properly the web UI 'sticks' horribly otherwise. I complement this by using ODBCGet in the dialplan. (Previously I #included dnd.conf, calldiversion.conf to achieve this functionality) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop
Leif Madsen wrote: I'm trying to load Asterisk, however I am getting the following error: [skipping res_musiconhold.so] [chan_sip.so]May 22 18:42:24 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_moh_stop May 22 18:42:24 WARNING[16384]: loader.c:421 load_modules: Loading module chan_sip.so failed! I've tried doing res_musiconhold.so=no in modules.conf with no change. This res is a requirement for current versions of chan_sip So, definitely *don't* have this in modules.conf: noload = res_musiconhold.so The question therefore is why is this res being skipped? Missing musiconhold.conf ? F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some problems with download Asterisk-addons
Fabio Donaggio wrote: I have some problems with the download of Asterisk-addons. [EMAIL PROTECTED] src]# export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot [EMAIL PROTECTED] src]# cvs login Logging in to :pserver:[EMAIL PROTECTED]:2401/usr/cvsroot CVS password: cvs [login aborted]: connect to cvs.digium.com(65.38.23.22):2401 failed: Connection timed out Looks like a Firewall problem to me - can you succesfully use CVS against other repositories? F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mysql
Fabio Donaggio wrote: I can't download asterisk-addons...I try with CVS, but i can't. How can I do??? http://asterisk.org/index.php?menu=download export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login - the password is anoncvs. cvs checkout asterisk-addons works for me... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codec used on E1
Bill Carroll wrote: I've got a * box with a TDM40B and an E100P. The E1 is connected to a Shout900 VoIP gateway from net.com. My question is, when I place a call from an analog phone and that call is routed over the E1, what encoding is used? And do I have any control over that? I'm obviously not an expert here, but the Shout900 vendor is claiming that some difficulties that we are having with integration are due to the use of ADPCM encoding on the E1 and he is saying we should be using A-Law or Mu-Law. I can't find anywhere that ADPCM is referenced as a codec that would be used in my situation or how I would set it if it was an option. A-law = alaw Mu-Law = ulaw Both are G.711: http://voip-info.org/wiki-ITU+G.711 F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G729 codec for asterisk
brian wrote: http://www.voip-info.org/wiki-Asterisk+G.729+licensing The Wiki is a bit wrong.. you can record raw g729 streams to disk, what do you think format_g729.c is? Fixed :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Psssst. The US is asleep - let's talk intern ationalization !!!
Olle E. Johansson wrote: Could we do it like this: languagecode[_countrycode][-fileset] Meaning languagecode: ISO two letter language code countrycode: ISO two letter country code fileset: up to 8 letter code for choosing a set of files within the lang code directory se-förför: Would look for files in the se/förför directory, use se syntax en_au-male: Would look for files in the en_au/male, use en syntax The language code controls the syntax used for saying phrases where we have such support The language code + _ + countrycode controls the directory All sounds good - nice flexible In some cases, the country code *may* affect syntax too. We had a case with english vs american english in say.c Apart from this. However, we can easily put in different syntaxes which match on the longer string, if required. I think this is flexible enough. And if you want to have a us texas voice, you can freely add it to en_tx, en_us-texmale or tx_north-female hehe ;) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Line appearances
Joseph wrote: I am trying to get an understanding of how line appearances work like on the cisco 7960 phones. Is there a wiki somewhere about how this works? General info on 79XX: http://voip-info.org/wiki-Asterisk+phone+cisco+79xx Also, the 7960 phones let you register more than one ext. Why would you want more than one or is this connected to line appearances? Yes Is there a way to have phones use more than one codec, say use g.711 to talk with * and g.729 to talk with another phone? No F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] German sound files available
ePyron Felix Deierlein wrote: But I am still not sure, where I sould place the german digits, letters and phonems. First I placed everything under sounds/de/.. but then digits did not work, then I linked it to /sounds/digits/de/ now I have german digits but saynumber is still english. I think you must have an older version of Asterisk. saynumber() now works for German in CVS HEAD I am currently working with Philipp to tweak this also to extend support to saydate() app_voicemail. The question where to place the subdirectories. In the wiki is not a real answer.. /var/lib/asterisk/sounds/de /var/lib/asterisk/sounds/digits/de /var/lib/asterisk/sounds/letters/de /var/lib/asterisk/sounds/phonetic/de F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7940/7960 SIP functionality questions
On Wed, 2004-04-21 at 20:20, David Carter wrote: I'm considering using Asterisk with some type of Cisco phone, and currently considering either the 7940 or 7960 because of its more-complete functionality (compared to the 7905). I'm currently wondering: Do all the expected functions (transfer, conference, voice mail, message waiting indicator, etc.) work normally with Asterisk over SIP? All work great :) What caveats are known about using these phones with SIP, as opposed to Cisco's proprietary SCCP? If an SCCP module is available for Asterisk, how functional is it? There are 2 SCCP modules chan_sccp chan_skinny I've not personally used either yet, but I believe they offer working basic functionality, but are not as advanced as SIP/IAX or, indeed, SCCP with CallManager. How customizable are the phone menus while using SIP (or if a SCCP module is available, using SCCP)? Services menu is very customisable: http://www.voip-info.org/wiki-Asterisk+Cisco+79XX+XML+Services There is even a manager interface for Asterisk available!: http://asterisk.edihost.co.uk/am-web/ Cisco doesn't seem to have much documentation online about using these phones in SIP mode, so if anyone is using these phones now, I'd appreciate hearing about your experiences. A good resource is: http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digits in a different language...
Fran Boon wrote: Carlos Chavez wrote: http://bugs.digium.com/bug_view_page.php?bug_id=0001097 This bug has been fixed in current CVS HEAD (not by using the patch in this BugID, though). I updated from CVS as you suggested but somehow things are worse now. Now ALL sounds are in english. I checked my configuration files and I have language=mx in my sip.conf and zapata.conf. I noticed that /etc/zaptel.conf has a defaultzone=us, could this be affecting the new version of Asterisk? It was working before I upgraded. Sorry, 'mx' isn't yet supported - it will default to 'en' Actually, looking at this again, 'mx' should still play digits from 'digits/mx' although the syntax followed would be the default 'en' syntax. I tested this all seems to work ok on my system. What is your directory structure? I think you must have another problem... Anyway, I have put up a patch to the bugtracker which adds gender supoport to saynumber() for 'es' syntax so that we can have 'la una' in a future saytime(). Also in this patch is setting 'mx' language to use the 'es' syntax. Please test if you get the chance: http://bugs.digium.com/bug_view_page.php?bug_id=0001566 Best Wishes, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digits in a different language...
Carlos Chavez wrote: Here in Mexico we use the same tones as in the US. In indications.conf I simply copied the [us] section and labeled it [mx]. In the general section I put country=mx. Since we do not share the same tones as Spain I thought it would be better to use a different setting than es. But maybe I got confused and I can set language=es and country=mx without there being a problem. I had this confusion when I started - 'uk' indications, yet 'en' language. By the way, is there a solution for saying numbers in spanish? The problem I have is that numbers like 100 (cien) do not sound correctly when concatenated to other numbers. When you have to say 101 the 100 sound changes to: ciento uno. ok, we have an 'es' syntax for saynumber() but it doesn't seem to support ciento uno as yet. Is this the only number that changes? What about 102? 110? 1001? In the current syntax patch, it requests a soundfile 'cien.gsm' as that's what the original patch author made available for download: http://voip-info.org/wiki-Asterisk+sound+files+international If you have a more complete set (this only includes digits) then could you make it available for download? I would prefer to rename 'cien.gsm' as the more standard '100.gsm' 'ciento' is basically 'hundred and', right? For other languages we standardise on '100-and.gsm' for that kind of thing... Comments welcome :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digits in a different language...
Carlos Chavez wrote: My sounds live in: /var/lib/asterisk/sounds/mx /var/lib/asterisk/sounds/digits/mx Until I upgraded yesterday to the latest CVS I got most sounds from the mx directories. I only had the problem with some digits. Since the upgrade all sounds play as en. I am still unable to replicate that here - all works good on latest CVS. (I copied my 'es' files to 'mx' to double-check they'd actually get played as well as the console saying it would attempt to play them) Do you have any custom patches? Do 'mx' files have the same names? The console shows they're requesting in mx or en? F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Digits in a different language...
Carlos Chavez wrote: ok, we have an 'es' syntax for saynumber() but it doesn't seem to support ciento uno as yet. Is this the only number that changes? What about 102? 110? 1001? All numbers like 10,20,30,40,50,60,70,80,90 and 100 have this problem. They all change when you have another number after. The only exception is the 1000 sound which does not change. For example: diez dieci (10) veinte veinti (20) From 30 to 90 you have to and an y (and) to the number. Some sounds like oh.gsm I simply recorded as cero (zero). Actually this is already in the current version. '100.gsm' is 'ciento' Hence the need for a separate 'cien.gsm' for simply '100' 11-19 21-29 have separate soundfiles with dieci/veinti 30-90 add 'y.gsm' before the last digit As I see it saynumber() appears correct to me. Now we 'just' need to work on saytime() saydate() http://voip-info.org/wiki-Asterisk+sound+files+international If you have a more complete set (this only includes digits) then could you make it available for download? I was waiting until everything is working. I have almost all sounds translated, just missing a few like ACD. I can make them available if you wish. As the usual Open Source motto goes - 'Release Early, Release Often' ;) Seriously - I for one would greatly appreciate it as we have many offices across latin america... NB My latest patch to get 'mx' language setting to use 'es' syntax has been accepted into CVS we also have the option to pass gender to saynumber() for 'una', playing 1F.gsm I will send an updated i18ntestsuite.conf for you to play with if you're working on this with me... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX Channel Capacity
On Sat, 2004-05-01 at 01:02, [EMAIL PROTECTED] wrote: -SNIP- With a IAX trunk, I have already observed (at the house) serious call/voice deterioration due to channel overload. How does one stop this? I.e. it would be very desirable to specify channel capacity (say xx number of simultaneous calls allowed) and force an 'unavailable' if more is demanded. incoming/outgoing limit: http://bugs.digium.com/bug_view_page.php?bug_id=849 There's a bounty on adding this important (critical!?) functionality michaelrose: are you still willing to fund this? If so, how much? Anyone else up for adding funds? (I might be, if I can get budget) diana: Are you willing to submit a patch to the bugtracker? Any information on this should be added here: http://www.voip-info.org/tiki-index.php?page=Asterisk+bounty+IAX+incoming-outgoing+limit F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Sat, 2004-05-01 at 16:42, Gavin Hamill wrote: Ah I've only noticed this thread has forked to CLID-in-non-Bellcore areas :) PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE PLEASE can some Kind and Worthy soul spend a little time in getting this really important feature implemented? You would have the undying gratitude of thousands of X100P users all round the world! :D Without CallerID support, the amount of 'cool stuff' you can do on a 1 line system is much reduced! :( (please?) How about setting up a bounty? http://voip-info.org/wiki-Asterisk+bounty F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk RedHat Enterprise
Asterisk wrote: Are their any issues with Asterisk and Redhat Enterprise? I have see one or two posts with issues concerning compiling zaptel drivers but that is about it. Just looking for some consensus to if any problems exist with it. Works perfectly for me :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk no card
Altus Snyman wrote: Is it possible to run asterisk and sip without any cards,(t100,voicetronix) Just a plain linux server,running mail and web, and add asterisk Yes F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Questions about alarm reporting in Asterisk
On Wed, 2004-04-21 at 18:41, Ernest W. Lessenger wrote: We use a package called Nagios to monitor our servers, which works quite well. It has the ability to track service and host dependencies so you don't get flooded with a bunch of service down alerts when the real cause is a bad switch (or similar). Nagios is great :) Here is some basic info on integration with Asterisk: http://www.voip-info.org/tiki-index.php?page=Asterisk+monitoring It would seem logical for someone (hah!) to write a res_snmp.c for asterisk that would expose a lot of asterisk's internal data. This would seem a logical step toward writing fully functional monitoring applications as well. The module would allow clients to add themselves to the list and receive traps, as well as check for the current status of various variables. brainstorming Okay, this may be over the top, but here goes. Write an asterisk application that sends (and receives) status information to another box over the PSTN. My idea is not only to use this as a way to verify that * is running, but as a way to RELIABLY tell that a remote * box is actively accepting incoming calls. It wouldn't have to be anything complicated, just a heartbeat and some basic details to let the caller know that yes, I'm alive and accepting calls over this line. Simplified protocol: 1) Monitoring box calls up and says (in DTMF): #my CallerID#extension I am trying to reach#I'm a machine, so reply in DTMF instead of voice#the secret code is# 2) The remote box says #your CallerID#Your DNIS#yes I will accept a call to that number# 3) Monitoring box acknowledges and disconnects 4) Remote box disconnects 5) Monitoring box decides whether it likes the answers it received and performs actions accordingly. /brainstorming Great stuff - I've added this the other comments to the Wiki page :) - please keep adding stuff there as it's an important area where we could benefit from sharing ideas ( implementations!) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Ser and Asterisk together
On Wed, 2004-04-21 at 21:02, AJ Grinnell wrote: Thanks, those are the advantages I needed to hear. FWD SipGate apparently have this config: http://www.voip-info.org/wiki-Asterisk+at+large Is there any special config I need to do to either * or SER? Do I just set SER as a friend in sip.conf? Still looking for documentation on using the two together. http://www.voip-info.org/wiki-Asterisk+config+sip.conf (See Example 2) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaprtc
On Tue, 2004-04-20 at 17:23, Steven Kokinos wrote: does anyone out there using zaprtc know how to go about initializing it at boot time? i have it compiled and working properly, but there is very limited documentation. Yup, works great for me :) Add this to rc.local to get it initialised at boot: insmod /lib/modules/2.4.21-9.ELcustom/misc/zaprtc.o /usr/local/bin/rtcsetup (Obviously modify the kernel path if required - this is for RHES3) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pattern matching rules for least cost routing
On Tue, 2004-04-20 at 23:21, Mark Elkins wrote:
-SNIP-
;Cell Phone call
exten = _00[78][234].,1,Playback(posix-cellphone)
exten = _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
;Default catch all - just dial it
exten = _0.,1,Playback(posix-defaultroute)
exten = _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
No matter what is dialled - I always go out on the 'Default' line.
Swapping order makes no difference. If I comment out the 'default' - it
does match the 'Cell' pattern - and works.
Pattern-matching within a context is not done based on order at all.
To achieve the effect you want:
include = cell
include = default
[cell]
exten = _00[78][234].,1,Playback(posix-cellphone)
exten = _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
[default]
exten = _0.,1,Playback(posix-defaultroute)
exten = _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
F
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Re: [Asterisk-Users] Does RTP traffic go through Asterisk IP PBX ?
PTCHEN wrote: Is there anybody knows if RTP traffic goes thru Asterisk IP PBX? If it is, it must limit the capacity of Asterisk. Do you know the concurrent SIP call capacity? And Is there any guy modify the source code to prevent this? Can be done already: http://voip-info.org/wiki-Asterisk+Letting+SIP+clients+connect+directly F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Database for extensions+vm+sip
On Sun, 2004-04-18 at 18:51, Carlo Pires wrote: Is database available only for sip friends ? Is possible to put voicemail.conf and extensions.conf into db ? Yes, all possible: http://voip-info.org/wiki-Asterisk+configuration+from+database F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Internationalisation/Internationalization
On Tue, 2004-04-13 at 16:11, Benjamin Wakefield wrote: On the tiki it says for international digits, I can dump them in the digits/au directory. I tried that -- just because, I also made a copy in au/digits. When the queue announces the position I it says: -- Started music on hold, class 'default', on SIP/11-5324 -- Stopped music on hold on SIP/11-5324 -- Playing 'dcsi/queue-thereare' (language 'au') -- Playing 'digits/2' (language 'en') -- Playing 'dcsi/queue-callswaiting' (language 'au') See that? The digits are 'en'! I can't work out why. Bug: http://bugs.digium.com/bug_view_page.php?bug_id=0001097 Patch listed there doesn't work for me, I'd be very happy to see a fix... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Upgraded to latest CVS, now no IAX1?
On Tue, 2004-04-13 at 20:13, Tor Houghton wrote: Well, I use IAX1 between the clients on the inside of the NAT to my local Asterisk, and IAX2 between the local Asterisk and my remote Asterisk. Previously (I have not tried yet with current version), when both clients and Asterisk used IAX2, the clients would communicate directly with remote Asterisk and so confuse my NAT firewall. In iax.conf, set: notransfer=yes That prevents IAX from transferring call to remote Asterisk, so it will stay in path. F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ZAPRTC question(s)
On Mon, 2004-04-12 at 17:37, Tony Mountifield wrote: Zaprtc is actually a *replacement* for the standard RTC module. It provides the same facilities, but includes extra parts for Zaptel use. -SNIP- All very interesting, thankyou :) The zaprtc.c code is based on the rtc.c from 2.4.20. I am running 2.4.22, so I isolated the zaprtc changes, and re-applied them to a copy of the rtc.c from 2.4.22. It works a treat. I've also enhanced rtcsetup to be a proper daemon. Any chance of sharing these changes somewhere? e.g. Wiki Thanks a lot, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Friends and MySql
Alex Lopez wrote: What is the difference b/w USE_MYSQL_FRIENDS=1 and USE_SIP_MYSQL_FRIENDS=1 Not sure ;) Am I to think that this replaces the entrys in sip.conf for the registering clients?? Yes If so, I am hosed as I cannot get a ATA-186 to register via MySql, but if I leave the config in sip.conf all is well. Could someone send me one record from their sipfriends table that works??? http://voip-info.org/wiki-Asterisk+sip+mysql+peers I see that there is no place to specify nat=yes, host=dynamic, etc. in the table, or am I just barking up the wrong tree. http://bugs.digium.com/bug_view_page.php?bug_id=0001086 F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Getting info about changes in CVS
On Wed, 2004-04-07 at 17:20, Eric Wieling wrote: There are several ways to know what changes in Asterisk's CVS. This URL http://asterisk.gnuinter.net/files/changelogs/ contains fairly up to date CVS changelog summary information. You can also sign up for the Asterisk-CVS mailing list at http://lists.digium.com/mailman/listinfo/asterisk-cvs Archives of the Asterisk-CVS mailing list are at http://lists.digium.com/pipermail/asterisk-cvs/ Any chance of adding this list to the GMane archive? For me browsing list archives via NNTP is *much* nicer than web interfaces... Thanks a lot, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extensions.conf sending calls to Cisco AS5300
On Mon, 2004-04-05 at 22:02, Brian Rathman wrote: I have my server configured to send to send all PSTN traffic to my Cisco AS5300 gateway via SIP. I use the following line in the extensions.conf file to accomplish this: exten = _NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],240,T) Unfortunately, when I removed the T from the end of the statement, the calls still complete, but they drop as soon as the called party answers the phone. I thought that the T had something to do with a timeout, but I have also seen documentation referencing that it allows * to stay in the middle of the call to determine if the customer use the # key, etc. I have not been able to find the detailed documentation that I was looking for on this subject. Can someone please direct me to this? Also it is my understanding, that if * stays in the middle of the call, I can not use the g729 codec without licensing from Digium. If this is the case, is there a way that I can use g729 in pass thru and still complete calls to the gateway? Any help would be greatly appreciated. Sorry, 'T' prevents pass-thru: http://voip-info.org/wiki-Asterisk+G.729+pass-thru F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CISCO 7940 and directory/services problem
Simon Brown wrote: I have quite successfully set up the Services button to work on the 7940 running SIP. I have a metric-imperial converter, a foreign exchange rate calculator, a calendar etc available to users. The XML is really fussy though. Could you share these example applications? Thanks, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
Nicolas Gudino wrote: http://sip.house.com.ar/operator Hi Nicholas, Agree with the other feedback - looks beautiful, the auto-refreshes are exceedingly smooth...definitely vindicates using Flash for client-side :) I also agree that more buttons would be very useful. (Although some of my labels get cut-off as-is, so I'd like a slightly smaller font even with current size) In fact I'll have so many that I think what I really want is the option to group them into different folders - ideally the user could even create their own folder! Aside from this, I note that the webpage states See at an glance: SIP registration status and reachability How does this work? I can't see any difference on my system between registered unregistered clients (makes a big difference for SoftPhones). I'd also like to have an option to disable the 'Talking to' part - in some situations this might be undesirable. Thanks a lot for the contribution - I would urge you to continue further :) Best Wishes, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Gnophone installation problems
Gavin Hamill wrote: I'm using Mozilla 1.7a installed from a tarball. The needed libraries are just there: You've answered your own question. You installed Mozilla from a tarball. RPM therefore doesn't know about it. You need to install a recent Mozilla RPM :) or use --nodeps F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New soundfiles from Allison posted
John Todd wrote: I've finally uploaded the newest (LARGE) list of sound clips in .gsm format to the bugtracker. Great thanks a lot :) Assume they'll make it into CVS tarball sometime soonish. Please see http://bugs.digium.com/bug_view_page.php?bug_id=985 for details and a full sound file list (and a tarball of the sounds in gsm format.) So, should we open up a new bug for the next set of requests? ;) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LookupCIDName from ODBC/MSSQL
Matteo Rancilio wrote: Is it possible to LookupCIDName from a unixODBC/MSSQL database? Not exactly what you're after, but possibly interesting to you: http://voip-info.org/wiki-Asterisk+cmd+LookupCIDname This uses Asterisk's internal database. If you want to store in an external database, then this will do the trick: http://asterisk.bkw.org/other/app_dbodbc.c Used in a similar way to: http://voip-info.org/wiki-Asterisk+cmd+DBput F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] who has German voice files ?
Thomas Haeger wrote: Wait a week and you can have german files from one of our customers, who wants to donate such files. Great :) Please could you make them available from the following webpage? http://voip-info.org/wiki-Asterisk+sound+files+international If anyone has Spanish or Portuguese, then that would make me very happy! Best Wishes, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729 passthrough notes (wiki fodder?)
John Todd wrote: I did some cursory searching on the list archives, and was not able to come up with this solution, so I'll summarize. Someone else should put this on the Wiki, since I am terribly lazy when it comes to web-ifying things. http://voip-info.org/tiki-index.php?page=Asterisk+G.729+pass-thru Thanks John for the legwork :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound Transfer and the # key
John Congdon wrote: I have applied the patch and restarted Asterisk. But it still only requires a single # to transfer. Did I possibly miss something? This is just to show that it was applied. [EMAIL PROTECTED] asterisk]# pwd /usr/src/asterisk [EMAIL PROTECTED] asterisk]# patch -p0 ../old_asterisk/doublehash.patch patching file res/res_parking.c Reversed (or previously applied) patch detected! Assume -R? [n] Apply anyway? [n] Skipping patch. 3 out of 3 hunks ignored -- saving rejects to file res/res_parking.c.rej Patch failed - this is what this output is showing. As Matt said the patch needs modifying to patch cleanly against the current version of the code... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Codecs [G.729]
On Tue, 2004-03-09 at 00:29, wrote: I'm looking for advice for codec that works best for asterisk. Anyone has real testing with all codecs, specially with G.729. I have tested with single call on few codecs that come with asterisk by using IPTraf and the rate as of below: ulaw 64 Kbps, sample-based Also known as alaw/ulaw 166kbits/sec alaw 64 Kbps, sample-based Also known as alaw/ulaw 167kbits/sec gsm 13 Kbps (full rate), 20ms frame size 66kbits/sec speex 2.15 to 44.2 Kbps n/a iLBC 15Kbps,20ms frame size: 13.3 Kbps, 30ms frame size 57.6kbits/sec G.729 8 Kbps, 10ms frame sizelicense Single call using which protocol? I would like to know more about your methodology... Have anyone test it with G.729? Please let me know. John Todd has tested IAX2 in 'trunking' mode with a variety of codecs , including G.729: http://voip-info.org/wiki-Asterisk+bandwidth+iax2 Note that your results show Speex as better than iLBC for a single call, which differs from John Todd's... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP - Receptionist
On Mon, 2004-03-08 at 23:19, [EMAIL PROTECTED] wrote: Monastery is neat as a monitoring tool. The console's we're talking about also let the user pick-up calls etc. Try this: http://astguiclient.sf.net/ F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] speex codec problem
On Tue, 2004-03-09 at 19:11, John Chester wrote: A call from a hardware phone using ulaw to an Xten phone using speex fails. When the Xten phone answers the call, Asterisk produces an endless stream of error messages: WARNING[311313]: codec_speex.c:167 speextolin_framein: Out of buffer space This continues until I shut Asterisk down. Try applying the .reg file found here to Xten: http://bugs.digium.com/bug_view_page.php?bug_id=133 This worked for me :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960
On Tue, 2004-03-02 at 06:35, Micke Andersson wrote: Does anybody know or have good examples of using all functions in a 7960 (SIP) http://voip-info.org/wiki-Asterisk+phone+cisco+79xx F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] having users in sql
On Tue, 2004-03-02 at 08:36, Micke Andersson wrote: If I want to have all my users (sip) in q mysql I've tried a few thingies.. but I didn't gett all the needed fields.. like nat, callerid, etc etc http://voip-info.org/wiki-Asterisk+configuration+from+database F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format
On Sun, 2004-02-29 at 09:18, Olle E. Johansson wrote: Olle's chan_sip2 introduces a 3rd possibility: Using templates autocreate peers for the majority of user options storing just the passwords in the MYSQL database. Combining this with MYSQL_FRIENDS, storing template= settings in a database would be very powerful. funnily enough, the application I'm writing at the moment has template= in the database :) However, I still need more per-user settings than is possible with MySQL_FRIENDS My own list of things that I need storing per user: extension shortname (for a mapping.conf that redirects [EMAIL PROTECTED] to an extension) fullname (for CallerID) email (for voicemail) accountcode pickupgroup template language calldiversion dnd Note that mapping.conf calldiversion/dnd need to modify files #included within extensions.conf as well as the user definitions. So I'd need a MYSQL_EXTENSIONS kind of functionality as well :/ All seems to be working nicely now, but I'm worried about getting time to 'restart when convenient' on a busy system - users won't want their calldiversion/dnd settings to only take effect overnight. I guess I need to implement this with astdb instead of MySQL, since this can be queried direct within the dialplan. Would be lovely to have dbget/dbput routines for MySQL as well as just db1! Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format
On Fri, 2004-02-27 at 21:39, Chad Sawyer wrote: In the contrib/scripts directory I have been trying to figure out the format of the entries in the MySQL table. -CUT- There are 3 different approaches to storing users in a database. The first is dynamic - the user details are read directly from the database. This is used for SIP IAX friends also for Voicemail: http://voip-info.org/tiki-index.php?page=Asterisk+sip+mysql+peers http://voip-info.org/tiki-index.php?page=Asterisk+voicemail+database However the number of options supported by this 'MySQL friends' system is currently very limited. The other possibility is to store all the details in the database when changes are made, write out new versions of the conf files. This is the approach taken by res_config: http://voip-info.org/wiki-Asterisk+res_config also by the contrib scripts, such as: retrieve_sip_conf_from_mysql.pl Obviously, the disadvantage of such systems is that Asterisk needs to be reloaded to see these changes, which can be disruptive to calls in progress, or may never get the chance to happen 'when convenient' on a busy system. Olle's chan_sip2 introduces a 3rd possibility: Using templates autocreate peers for the majority of user options storing just the passwords in the MYSQL database. For me, the ideal would be to hack the code to extend the functionality of MySQL friends...however I'm not a C programmer. I am currently starting work on the 2nd option since I want to expose as many options within a web-based GUI as possible. Any suggestions on how to minimise impact on the running system during reload are welcome :) Best Wishes, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] retrieve_sip_conf_from_mysql.pl data format
On Fri, 2004-02-27 at 21:39, Chad Sawyer wrote: In the contrib/scripts directory I have been trying to figure out the format of the entries in the MySQL table. It isn't at all obvious is it? I've now worked out what it does have written this up on the Wiki, along with my previous post about database integration in general: http://voip-info.org/tiki-index.php?page=Asterisk+sip+conf+from+mysql http://voip-info.org/wiki-Asterisk+configuration+from+database Now back to the task of getting a workable UI for my specific situation's needs ;) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] exit
Greg Kedrovsky wrote: You must have started asterisk with asterisk -c No, I started it with asterisk and had it running in the background. Suggest starting as 'safe_asterisk' asterisk -r exit Always works for me... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail not working with mysql!!!!
atif wrote: I need some tips on configuration of voicemail with mysql... http://voip-info.org/tiki-index.php?page=Asterisk+voicemail+database here is my voicemail.conf **voicemail.conf*** [general] dbhost=localhost dbname=asteriskvmusers dbuser=root I assume you just removed the line: dbpass=password for display on email? I have created the databaseasteriskvmusers in mysql and then created the table 'users' in that database. but it's not working...i mean when I change the passward through the zap interface it is changed in the file 'voicemail.conf' but database is not effected at all... Did you compile Asterisk with USE_MYSQL_VM_INTERFACE=1 ? one more thing which one is newer versionand has mysql support voicemail or voicemail2 'voicemail' is deprecated. When people talk about voicemail these days, they mean 'voicemail2' F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail not working with mysql!!!!
Tim Sailer wrote: one more thing which one is newer versionand has mysql support voicemail or voicemail2 'voicemail' is deprecated. When people talk about voicemail these days, they mean 'voicemail2' Really? In the docs somewhere, it shows voicemail2 as deprecated. ok, 'voicemail2' has been renamed as 'voicemail' F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pickup
Jim Sneeringer wrote: The extension for Pickup seems to be *8#, but I cannot find it anywhere in any configuration file. Is this a hard wired extension? Yup, you can edit it in the source code recompile. I set mine to '**3' to fit the legacy PBX system: vi res/res_parking.c +54 static char pickup_ext[AST_MAX_EXTENSION] = **3; F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec Order / Preference
Eric Wieling wrote: You cannot specify the order of codec selection with Asterisk My understanding is that when using SIP, the order within the [general] section does affect priority. This has been confirmed by my own testing. However the order within individual user/peer settings, don't. I see this is on the wishlist for chan_sip2. I don't have so much experience with other channel types. F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users] Re: [Asterisk-Users] SIP config documentation
On Sat, 2004-02-21 at 09:06, Costa Tsaousis wrote: callgroup= ; UP pickupgroup= ; UP Q4: Since a user cannot accept calls, why to setup call pickup for him/her? Sorry, haven't used or checked call groups. Anyone else? No answer on this yet... I use pickup groups just fine (using type=friend). I agree that it only makes sense to me for Peers...I can't speak for why the code is currently the way it is. Perhaps logging it as a BUG would be a better way to draw comment? accountcode= ; U- CDR's account code incominglimit= ; U- concurrent call limitations ( = 0 ) outgoinglimit= ; U- concurrent call limitations ( = 0 ) Q6: How is it possible for a type=user phone to have BOTH incoming and outgoing limits? Interesting question. Anyone else? No help on this either so far. I don't yet have a need for the feature can't comment on the reasoning for the current settings in the code. However your logic seems right to me: incominglimit should be for peers outgoinglimit should be for users It looks to me like a BUG: Maybe you should log it as such (along with a little patch if you can!) mask= ; -P netmask for host= parameter. This has to be defined *before* the host= parameter. Thanks for the hint. I didn't notice this. What it does? Don't know. Anyone else? Why do Asterisk apply a host mask to an IP address for a host? This is still open too. Since this is for peers (i.e. outgoing calls) we need to be more precise than a subnet for directing the calls out, so it can't fully replace username/password (unlike the simple host=, which can) It only makes sense to me for providing additional security restrictions to the username/password for the REGISTER to be succesful. I can't comment on whether this is how it gets used in the code. F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP config documentation
On Sat, 2004-02-21 at 11:48, Olle E. Johansson wrote: canreinvite has yes|no|update as keywords. with UPDATE a SIP method UPDATE is initiated to change the media path. with YES, a new INVITE is issued within the current call. (a re-invite) with NO, the call stays within asterisk. Any ideas on when UPDATE would be better than the standard re-invite? callerid= ; U- caller id of the user: Name number. Have to check this one. Been working a bit on this problem in the chan_sip2 channel. I have submitted two bug reports. One includes a patch to chan_sip.c that fixes the problem. See: http://bugs.digium.com/bug_view_page.php?bug_id=0001074 Another is about CALLERIDNUM. This variable seems to strip the dots from the domain without practical reason (also SetCIDNum and the like do this). See: http://bugs.digium.com/bug_view_page.php?bug_id=0001075 I'll add comments to the bug reports. We really need to think about how we steer the SIP channel forward. Asterisk is a multi-protocol PBX, so there's no sense in making the SIP channel a stand-alone SIP proxy that doesn't work with the rest of the PBX. If you don't want a PBX in the core of your SIP network, use a SIP proxy there and use Asterisk where it fits in. This way of reasoning means there are things that will never happen in the Asterisk SIP channel because of the multi-protocol architecture. We need to be clear on that while moving the channel forward. Could different options be used in different channels? i.e. if the SIP user is calling a Zap channel, then you'd use only standard numerical CallerIDs. However, if the SIP user is calling another SIP user, then these new patches would kick in you'd be able to have full SIP URLs viewable :) Having said that, there's a lot of things to do to make the SIP client and server within Asterisk more compliant and functional with the rest of the SIP world. Yes, please :) I do want Asterisk to be in the forefront in preventing use of Asterisk as a open SIP spam relay to use mail terms. Mail servers are picky of which domains they server for inbound and outbound messages, in some cases also on what domain is used for outbound messages. We need to have configuration that follows this line of thinking for SIP. If someone is using our domain for outbound calls - authenticate. If someone is randomly placing calls to extensions in any domain they invent *into* our PBX, don't answer. This is good thinking - the idea of VoIP spam fills one with horror! As I stated earlier, I'm highly suspicious to the in order of preference part. Since I got no comments or replies on that mail, I suspect I'm right :-) What do you mean? Is the order of preference not working? I've checked into this since no one else bothered... :-) Allow/deny codecs in the [general] section have an order of preference, it is correct. Allow/deny codecs for peers/users have no order of preference, it's just a matrix of codecs we can use for calling them. This looks like a BUG to me! Is it only evident in the SIP channel, or in others (such as IAX) as well? I can't find it in the BugTracker - is it there? Sounds easy to fix, but that's easy for me to say, since I'm not a coder ;) A new question: Is chan_sip2 ready for production? I'm using it in my production servers, but I wouldn't recommend it to anyone else for more than testing at this time. It's beta and I'm adding new stuff constantly, propably new bugs as well. Have got some very useful feedback (thank you, Fran!) but need much more from testers. It adds a lot of features, like templates and MYSQL authentication. Adding features is dangerous without testing, so please help me test this little creature. I'm testing Asterisk in a SIP-only environment using this version of the channel. I don't yet run a full Production system, but test this with X-Lite, ATAs 7960sworks great so far :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users] Re: [Asterisk-Users] SIP config documentation
On Sat, 2004-02-21 at 09:06, Costa Tsaousis wrote: incominglimit= ; U- concurrent call limitations ( = 0 ) outgoinglimit= ; U- concurrent call limitations ( = 0 ) Q6: How is it possible for a type=user phone to have BOTH incoming and outgoing limits? Interesting question. Anyone else? No help on this either so far. I just managed to find some stuff in the bugtracker (weird - no hits, then browser crashes hits show upon restore!) Original patch: http://bugs.digium.com/bug_view_page.php?bug_id=098 Not working as expected: http://bugs.digium.com/bug_view_page.php?bug_id=329 More work done to it: http://bugs.digium.com/bug_view_page.php?bug_id=408 1 thing that I certainly wasn't expecting is the directions mentioned: Incominglimit = number of calls the local extension can originate to Asterisk. Outgoinglimit = number of calls Asterisk will terminate to local extension. The fact that isn't working for peers yet is clearly mentioned. I'm not totally sure whether it now works for non-local users or not. F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP config documentation
On Sat, 2004-02-21 at 19:30, Costa Tsaousis wrote: I believe there are three possible paths for asterisk: 1. Stick to the switched world (as the common denominator for telephony). This means that * can have any number of gateways on it, but always, it will be a switching-like PBX with some VoIP functionality, build in software. Example: forget about SIP, H.323 and the like, focus on switched telephony with the ability to place and receive calls via VoIP making them as switching-friendly as possible, of course with limitations. 2. Open up the possible scenarios and let the administrator choose the primary protocol. This means that although asterisk will not address in its core artificially intelligent thinks such as callerid convertion, it could be configured to follow either schema. In this case, the administrator should have the means to configure some aspects of the secondary protocols. Example: Support a number of protocols as primary and configure asterisk according to them. This means that various building blocks (like callerid handling) will be multiplied, one variation per primary protocol and the whole system will support only one protocol for such functions. When there are interfaces with other protocols the administrator should make hard decisions about the properties that cannot be converted automatically. This is my case today. I want SIP as primary and I know I will have to provide numeric callerids in the configuration when interfacing with other protocols. What I need is a SIP PBX and IVR, not a SIP proxy. I cannot do what I need with SER (at least not that easy...). This is also my situation. 3. Build asterisk as a superset of all protocols, internally. In our case this could mean that the callerid could be defined as: callerid=TEXT internet address number or even: callerid=Your Name sip:[EMAIL PROTECTED] h323:... mgcp:... or just provide directory services for callerid convertions between non compatible protocols. As a superset of all protocols, asterisk will be able to be a fully functional member of each of the supported worlds and will be able to also handle all the protocols as primary at the same time. Path 3 is the perfect path. Path 2 is a good one. Path 1 demotes asterisk. If it is going to be Path 1 I believe we, all, are going to use asterisk as a secondary component in our telephony infrastructures that will provide some valuable services, but it will never be the heart of it, unless all that we need is a plain old switched-like PBX with some VoIP functionality. I agree that these are the possible paths would personally be quite happy with (2) if the incompatibilities are well-documented. We need to work together to handle the multi-domain scenario. Please send me whatever you have and let's continue discussing this so we get a solid architecture. I do want Asterisk to be in the forefront in preventing use of Asterisk as a open SIP spam relay to use mail terms. Mail servers are picky of which domains they server for inbound and outbound messages, in some cases also on what domain is used for outbound messages. We need to have configuration that follows this line of thinking for SIP. If someone is using our domain for outbound calls - authenticate. If someone is randomly placing calls to extensions in any domain they invent *into* our PBX, don't answer. I agree with you, but still I think all these should be configuration decisions, not implementation ones. Yes, of course - in a purely internal network, you may legitimately wish to run an open relay. However, the default settings (*.conf.sample) should be configured safely if at all possible... F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP config documentation
On Wed, 2004-02-18 at 16:06, Arretni VoIP Tech wrote: Can musiconhold=class be included in sip.conf? I want to play music on hold for calling users on the VoIP side. Currently, I can only play moh when the call came from the PSTN (zapata). Use Olle's chan_sip2: http://bugs.digium.com/bug_view_page.php?bug_id=759 F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] running asterisk as non-root
Due to security reasons I want to run asterisk as a non root. http://voip-info.org/tiki-index.php?page=Asterisk+non-root This HOWTO works for great for me :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Speex == Screech using version 1.1.4
Florian Overkamp wrote: I am using X-Lite on some setups. Speex from X-Lite does not seem to work with asterisk - I just get no sound at all. Disabling Speex and favouring GSM or G711 works fine. Need to apply a .reg file to the PC running X-Lite: http://bugs.digium.com/bug_view_page.php?bug_id=918 F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.2 RPMS Updated
Brian West wrote: Is someone going to do the v1-0-stable RPMS? Not sure if anyone knows that it was branched yet or not. Everyone was jumping up and down and chanting BRANCH BRANCH BRANCH I don't think that we've reached 1.0 stable, though, have we? branching is an essential precursor in order to allow stablisation of the current featureset to happen in a different space to the addition of new features. Personally I welcome this - both branch HEAD should benefit :) However, I think it's too early for RPMS of a snapshot of this branch of CVS ;) It finally happened and nobody says a word haha.. :) I didn't see any announcement ;) My word: Thankyou :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as non root
On Thu, 2004-02-05 at 14:03, Chris Lee wrote: I followed the wiki instructions: http://www.voip-info.org/wiki-Asterisk+non-root Glad someone's finding it useful :) Now I have a working asterisk running as user asterisk. I do however have some problems: 1: I dont have access via asterisk -r root should have access using asterisk -r (does for me anyway) 2: The pid file is no longer being updated If this is an upgrade to a previous install, then check /etc/asterisk/asterisk.conf to see whether the change to ASTVARRUNDIR has taken effect in the config file... 3: I want to create a file in init.d so that I can use service start and stop, but need to be able to pass asterisk the gracefully command etc, any ideas welcome. maybe: asterisk -rx stop gracefully etc pass F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with big number of extentions.
On Sat, 2004-01-31 at 10:36, WipeOut wrote: Fran Boon wrote: OK, so what success have people had with which clustering technologies? I'm more interested in resilience than performance. I would think that failover clustering would be far easier than a load sharing or processing cluster.. Great, so that works for me :) For lots of info on various clustering a HA systems take a look at http://www.linux-ha.org/ This looks like a great resource :) Has anyone successfully used this with Asterisk? Cheers, F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Words for Allison(?)
On Sat, 2004-01-31 at 18:24, Rob Fugina wrote: In the mean time, I've seen references to bug #'s, here on the list and in the CVS logs. I've yet to stumble across the bug tracking system, though -- can you give me a nudge in the right direction? http://bugs.digium.com/ F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk with big number of extentions.
Anton wrote: you can do it with a well setup cluster OK, so what success have people had with which clustering technologies? I'm more interested in resilience than performance. Thanks a lot, F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X-Lite Asterisk: Speex iLBC not working?
This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing iLBC, there is some very garbled noise, but nothing intelligible. Sniffing the packets, I can see that X-Lite Asterisk have chosen differing 'Payload type' numbers: X-Lite: a=rtpmap:97 speex/8000 a=rtpmap:98 iLBC/8000 Asterisk: a=rtpmap:97 iLBC/8000 a=rtpmap:110 SPEEX/8000 According to the Speex RFC, this is acceptable: http://speex.org/drafts/draft-herlein-speex-rtp-profile-00.txt Dynamic payload type codes MUST be negotiated 'out-of-band' for the assignment of a dynamic payload type from the range of 96-127. I'm wondering whether the system is at all case sensitive? From the RFC: When conveying information by SDP [4], the encoding name SHALL be speex. NB Ethereal shows payload-type as being 97 when X-Lite reports iLBC 110 when X-Lite reports Speex, so the Asterisk numbers seem to 'win'. Any light shed on this would be great. Whilst GSM is ok, it would be great to leverage the power of Speex :) Thanks a lot, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN incoming - both SIP H323 always arrive in default context :-?
Some of you may remember seeing my issue using SIP for incoming calls from the PSTN: http://voip-info.org/wiki-Asterisk+cisco+FXO i.e. all incoming calls arrive in the default 'bogon-calls' context. Well, I tried again using H.323 get exactly the same result (both for chan_h323 chan_oh323) i.e. all attempts to put a type=peer in sip.conf or a type=user in h323.conf for my host are ignored/bypassed. Is this a bug? Luckily for me, I can firewall off the H.323 port to all bar this one IP, so I now have a workable solution...until I want to extend the H.323 gateway to other devices... Anyone get host=x.x.x.x to be able to bypass the default contexts with either SIP or H.323? Cheers, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 0.7.1 RH 7.3 RPMS Released
On Thu, 2004-01-22 at 18:17, Jonathan Moore wrote: I am researching the use of White Box Enterprise Linux. Someone else in a similar position with a bunch of 7.x boxes created it. He took all the SRPM files for REL v3 and removed all the Red Hat logos and trademarks. It is the same software as Enterprise but you can freely copy it. They also modded the update scripts to work with more generic update sources. The cool thing is the system is completely compatible with the REL source errata which Red Hat has promised to continue updating for at least five years. They have also setup a small system of mirrors to host the update files. It looks very promising. I am trying this and Debian to see which will be easier to keep updates for. Info and ISO file available at http://www.beau.org/~jmorris/linux/whitebox/index.html Someone else has a similar project going but it didn't seem to be as far along. Personally I think that Tao Linux (http://taolinux.org/) is better set-up than Whitebox. Anyway, for me, RedHat Enterprise Linux it's 'rebuild' clones are the best development target to aim for - this is the stable environment (5 years) that people like from 7.3, brought up to date supported by hardware manufacturers, software developers hence corporates. I'm running 0.7.1 on RHEL3 without any problems so far :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco FXO as PSTN gateway (updated request for assistance)
Olle E. Johansson wrote: I have been compiling information on this configuration onto the Wiki: http://voip-info.org/wiki-Asterisk+cisco+FXO I can call out to the PSTN just fine, but inbound calls all appear in my default [bogon-calls] context, not in [pstn-incoming] As I understand it, the Cisco is not registred with Asterisk as a peer. It /appears/ to be: redcusr01*CLI sip show peers Name/usernameHost Mask Port Status 1001/100110.129.3.128(D) 255.255.255.255 5060 Unmonitored PSTN 10.129.3.254 255.255.255.255 5060 Unmonitored Could you please mail a SIP DEBUG output of an incoming INVITE from the Cisco to Asterisk? redcusr01*CLI SIP DEBUG SIP Debugging Enabled redcusr01*CLI Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.129.3.254:5060 From: sip:10.129.3.254;tag=23D83A04-449 To: sip:[EMAIL PROTECTED] Date: Sun, 07 Mar 1993 23:02:53 GMT Call-ID: [EMAIL PROTECTED] Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 3055577250-442896844-2154029736-727233534 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: sip:10.129.3.254;party=calling;screen=no;privacy=off Timestamp: 731545373 Contact: sip:10.129.3.254:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 247 v=0 o=CiscoSystemsSIP-GW-UserAgent 7818 6792 IN IP4 10.129.3.254 s=SIP Call c=IN IP4 10.129.3.254 t=0 0 m=audio 19058 RTP/AVP 0 101 c=IN IP4 10.129.3.254 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 20 headers, 11 lines Using latest request as basis request Sending to 10.129.3.254 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found description format PCMU Found description format telephone-event Capabilities: us - 4, them - 4/0, combined - 4 Non-codec capabilities: us - 1, them - 1, combined - 1 Looking for 1001 in bogon-calls list_route: hop: sip:10.129.3.254:5060 Transmitting (no NAT): SIP/2.0 100 Trying All guidance very much welcomed :) Other options I'm considering to fix this are: (1) Using SER to take the incoming calls from the Cisco (2) Using H.323 to take the incoming calls from the Cisco Commetns on these 2 options also welcomed :) Best Wishes, Fran. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
