[Asterisk-Users] Codecs and more analog lines?
Hi! Are the GIPS codecs now implemented with the Asterisk? If I need more analog lines, say around 30, what's the easiest way doing it? I checked the Mediatrix box with 24 connections, maybe that would be a good (and rel. cheap) way to go? Any other suggestions? The ports has to support fax machines. rgds, /staffan -- -- Staffan Kerker / KIT Communications, AerotechTelub mail: [EMAIL PROTECTED] "Don't get involved in politics man, just play the gig..." /Sgt. Floyd, Electric Mayhem Band ___ 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
SV: [Asterisk-Users] Mailing list growth
Hi Isn't this exactly what we _don't_ wanna do?! =) I suppose TDM and VoIP is supposed to interconnect not to be separated. i think it's nice with a busy list, it means some real hot stuff is happening, and that's good! rgds /staffan -Ursprungligt meddelande- Från: Luciano Ramos [mailto:[EMAIL PROTECTED] Skickat: den 9 januari 2004 14:12 Till: [EMAIL PROTECTED] Ämne: RE: [Asterisk-Users] Mailing list growth What about splitting asterisk-users in 2 groups asterisk-tdm and asterisk-voip -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de Philipp von Klitzing Enviado el: Viernes 9 de Enero del 2004 08:52 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Mailing list growth Hi! > I still think we need something more fine grained. I think we can add > the asterisk-biz list, and eventually something akin to a newbie list, > but need a more appropriate name, IMHO. As discussed before that newbie is very likely to not work out; but here's a practical suggestion: Split asterisk-users into by introducing a "higher-level implementation" list that deals specifically with channelbanks & T1 issues (=larger installations). VoIP will remain on asterisk-users. That way newbies can stick to asterisk-users without the added traffic of asterisk-t1 (just a suggestion for a list name), whereas more experienced users still subscribe to both lists, but at least get a nice filter mechanism helping to decide what to read first. Cheers, Philipp ___ 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
[Asterisk-Users] Another * crash
I have an interesting problem now. I use asterisk to connect to both FWD and a sip provider here in sweden. suddenly, (i know my provider upgraded from a unknown SIP proxy to SER) Asterisk crashes when I try to make a call using this provider. FWD still works fine, and I can call directly towards the GW to POTS without any problems. But, as I call using my providers SER, Asterisk crashes. When I debug sip I get a "noisy feedback" from SER, and then asterisk crashes. the only debug information asterisk is leaving is "segmentation fault, dumping core". anyone got a clue? i'm not running the latest CVS so maybe i should upgrade first... rgds, /staffan kerker -Ursprungligt meddelande- Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Skickat: den 30 november 2003 20:04 Till: [EMAIL PROTECTED] Ämne: Re: [Asterisk-Users] asterisk server crashing >From the console, I see where the call comes in and I can see where the party from the outside hangs up. The next thing that is said is as follows: "libgcc_s.so.1 must be installed for pthread_cancel to work". Now I've taken a look on my system and I do in fact have the libgcc_s.so.1 on my system. The location is as follows: /lib/libgcc_s.so.1 It is part of the libgcc-3.2.2-5 package that I have installed on my system. I'm not a programmer, just a novice so I'm not quite sure how to run a backtrace or where the core file would be located. Thanks for your help so far. AJ On Sun, 30 Nov 2003, Roy Sigurd Karlsbakk wrote: > - What's the console output after the crash when starting asterisk with > -gvvvc? > - After the crash, run a backtrace of the core file and send the output > here > > ...perhaps this should be on the FAQ? > ...and perhaps the FAQ should be linked to from asterisk.org? > > roy > > On Sunday, Nov 30, 2003, at 14:14 Europe/Oslo, > [EMAIL PROTECTED] wrote: > > > I deleted all the asterisk related directories and their subdirectories > > from /usr/src/ and did a brand new check out of zaptel, zapata, libpri, > > asterisk-addons and asterisk. > > AJ > > > > > > On Sat, 29 Nov 2003, Tilghman Lesher wrote: > > > >> On Saturday 29 November 2003 20:36, [EMAIL PROTECTED] > >> wrote: > >>> Quoting [EMAIL PROTECTED]: > In the zaptel zapata and libpri directories I executed a make clean > and did a cvs update and then ran make install. In the asterisk > directory I did a make clean, a cvs update and a make upgrade. So > I guess the answer to your question is yes I did take care of the > other things as well. At least as far as I can see and as far as I > know. > AJ > >>> > >>> I don't know if your situation is the same as mine but I have been > >>> burned in the past by assuming that cvs update will provide all the > >>> lastest files. It only updates files that have previously been > >>> downloaded, soo, if you do not have a file that is now part of > >>> zaptel for instance, you will still not have that file. Do a fresh > >>> checkout to make sure you have all of the needed files. By the way, > >>> zapata is no longer needed. It has been incorporated into one of the > >>> others. > >> > >> Perhaps you mean subdirectories? True, 'cvs update' will not > >> typically > >> create new subdirectories, so you can do a 'cvs update -d' to have the > >> update create new subdirectories, as 'cvs checkout' does, but 'cvs > >> update' should create new files (in existing directories) just fine. > >> > >> -Tilghman > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIMPLE support in Asterisk?
Hi this SIMPLE support things sounds really good. too bad i'm not a programmer otherwise i would most certainly try to fix this. this together with a full SIP proxy possibility (or at least SIP URI handling) would be great! anyone else interested in this? anyone with programming knowledge interested in this? =) rdgs /staffan On Fri, 2003-11-28 at 08:51, Olle E. Johansson wrote: > Leif Madsen wrote: > > > On Thu, 2003-11-27 at 12:03, Mark Spencer wrote: > > > >>>Yea, cause I used both Kphone and Windows messenger, and they > >>>successfully registered (and subscribed i think) towards asterisk. Using > >>>Kphone I even get a online status on all other users on the asterisk but > >>>no interaction with status or IM. So maybe there is some quasi presence > >>>avaible? I think it would be a great tool to support IM/Presence. There > >>>is so much that can be done with such implementations. > >> > >>SIMPLE could be added within chan_sip, but there is no mechanism within > >>Asterisk to move text from one channel to another *without* the context of > >>a call. *With* the context of a call, we definitely have such a thing > >>(TEXT frames) > Some brainstorm notes then: > > So what's a "call" for asterisk? > * Something that's set up between two endpoints through the dialplan. > > Simple can send messages within a call, like > * A calls B with SIP (INVITE-ACK-ACK) > * B sends a URL to A with SIMPLE within the SIP session > > The problem that we have, if I understand Mark, is that Simple may also > be used to send IM without setting up a SIP call (INVITE-ACK-ACK). Like > the MWI SIP Notify message from Asterisk. Is that only generated in the > relation to a SIP register? > > Asterisk has some notion of presense (CLI> SIP show peers) but not detailed > as the normal IM user wants: Presence with some attribute (atoffice, athome, > atmistress etc). > > To get SIMPLE to work within Asterisk, we'll have to: > * Add SIMPLE support within the context of a call > * Add a new session apart from a "call" - "notification" > * Add some attributes to presence structure > * Add a SUBCRIBE/ACCEPT mechanism - who may subscribe to my whereabouts? > * Find programmers that can do this :-) > > Do the other protocols, MGCP, IAX2, H.323 have any support for text messages? > > > /O > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Staffan Kerker AerotechTelub AB, Communications [EMAIL PROTECTED] ph. +46(0)47042185 cell. +46(0)705391365 -- "Don't get involved in politics man, just play the gig... " /Sgt Floyd, Electric Mayhem Band ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIMPLE support in Asterisk?
Ok Yea, cause I used both Kphone and Windows messenger, and they successfully registered (and subscribed i think) towards asterisk. Using Kphone I even get a online status on all other users on the asterisk but no interaction with status or IM. So maybe there is some quasi presence avaible? I think it would be a great tool to support IM/Presence. There is so much that can be done with such implementations. rgds, /staffan kerker On Thu, 2003-11-27 at 04:53, John Todd wrote: > >Hi > >Is there any work being done on implementing IM/SIMPLE support > >for SIP on Asterisk? Like a presence server? > > > >rdgs, > >/Staffan Kerker > > No. > > There are currently requests in the system for that functionality > (http://bugs.digium.com/bug_view_page.php?bug_id=134) but it's > waiting for a White Knight to ride up and code a solution. > > However, there are some quasi-presence tools that appear to be built > into Asterisk in ways that nobody has explained yet. I wouldn't use > the term "secret" but the total lack of documentation and/or answers > to the questions on how to use these features makes me wonder... > > JT > > > > >Date: Thu, 16 Oct 2003 03:51:01 -0500 > >To: asterisk-users-lists.digium.com > >From: John Todd <[EMAIL PROTECTED]> > >Subject: Use of the "hint" modifiers - examples, anyone? > >Cc: > >Bcc: > >X-Attachments: > > > > > >I have found some references to the "hint" (or HINT?) variable and > >method in the source code, but quite a bit of Google-ing has not > >turned up any extensive answers as to some real-life examples of how > >to use this perhaps very useful tool. I understand the point of the > >tool, but I need to get some actual configs to look at before I > >think I'll figure it out. Even if my particular equipment doesn't > >support it, there may be other ideas I can get from it. (JerJer - > >maybe SCCP could use that data if there is an SCCP command of > >similar nature to the SIP SUBSCRIBE command - that would be pretty > >handy for those 7914 operator stations.) > > > >Searching through the source gives tantalizing hints (no pun > >intended) in pbx.c, but no actual real-life samples. Can someone > >who is familiar with it put some words to the features? > > > >I found this from March 20, 2003 from Andre Bierwirth: > > > > > >Subject: [Asterisk-Dev] Logged in users > >To: [EMAIL PROTECTED] > > > >I am currently work on it. If i am ready Asterisk have functions to get = > >device or extension state. > > > >int ast_extension_state(struct ast_channel *c, char *context, char = > >*exten) > >returns=20 > >-1 =3D error or no hint(device hint) for extension > > 0 =3D extension is free or unknown > > 1 =3D one device in extension is busy (have a call) > > 2 =3D all devices in extension unavailable(unregistered) > > > >** You can give ast_device_state a Dialstring like SIP/mark or IAX/mark = > >** > > > >int ast_device_state(char *device) > >returns > >-1 =3D error > > 0 =3D device is free or unknown > > 1 =3D device is busy (have a call) > > 2 =3D device is valid but unregistered > > > >So SIP can support SUBSCRIBE requests, and for Snom200 SUBSCIBE Dialogs = > >(Map a Key to an extension and see if the extension have a call (the LED = > >turned on)) > > > >Its easy to implement the device state support for IAX, i have talk with = > >mark about it. I implement only the PBX and Channel and SIP functions. > > > >With IAX you can poll the dialplan and get the extension states if its = > >implemented. > > > >Andre > >--- > > > >JT > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Staffan Kerker AerotechTelub AB, Communications [EMAIL PROTECTED] ph. +46(0)47042185 cell. +46(0)705391365 -- "Don't get involved in politics man, just play the gig... " /Sgt Floyd, Electric Mayhem Band ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIMPLE support in Asterisk?
Hi Is there any work being done on implementing IM/SIMPLE support for SIP on Asterisk? Like a presence server? rdgs, /Staffan Kerker ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and SIP Proxy on same machine?
Hi Is it possible (or recommended) to run both Asterisk and say SER on the same physical machine? How about port conflicts? Maybe the easiest way is to change the default SIP port on Asterisk? But how will that work if I register some SIP accounts directly from asterisk (like my SIP provider) but then wanna dial outbound pure SIP calls via my SER... Has anyone got a functional system like this up and running? SER/Asterisk on the same machine? rgds, /Staffan kerker -Ursprungligt meddelande- Från: CW_ASN - Gus [mailto:[EMAIL PROTECTED] Skickat: den 6 november 2003 12:45 Till: [EMAIL PROTECTED] Ämne: Re: [Asterisk-Users] How to control dialout in extensions file You could use DISA app. exten => 2101,1,DISA,/opt/pass.txt|default Where: /opt/pass.txt is a plain text file with password list. default is a destination context. Anyway, please do 'show application disa' from CLI. Hope this helps, Gus - Original Message - From: "Jacky Chen" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, November 06, 2003 8:18 AM Subject: [Asterisk-Users] How to control dialout in extensions file > Hi, all > > I have builded a pbx server for pstn, sip & h.323 users > but i can't find any example extensions.conf for access > control when users which call longdistance with pstn, > > If anyone have good example, please sharing your experience > Thanks very much > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] One more QoS question for RH9
Hi I know this is a bit off topic, but still pretty interesting. I'm running Asterisk on my Linux router/NAT/FW connected via cable (1mbit/200kbit) to the internet. Now, I wanna do local QoS implementation. Just very simple to give RTP (UDP) highest priority on my outbound interface. So, whenever I got an ongoing call, the RTP traffic should be handled first and other data (file transfers that may be ongoing) should back off. I've done this very simple with Cisco IOS routers by just using the priority queueing... but how do I do this (the easiest way) on a RH9 machine? Don't need any classes more than 1. priority RTP 2. all other data. rgds, /staffan kerker -Ursprungligt meddelande- Från: Béasse Christophe [mailto:[EMAIL PROTECTED] Skickat: den 31 oktober 2003 09:54 Till: [EMAIL PROTECTED] Ämne: [Asterisk-Users] Password in VoiceMail Hi, In voicemail. i declare : 1000 => 1234, ,[EMAIL PROTECTED] I don't see what's password 1234 is for ? password for what ? Where this password is used ? Where this password is defined ? Thanks Chistophe ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
SV: [Asterisk-Users] Running Asterisk and NAT on the same box?
Hi I'm running exactly the same setup. Asterisk is running on my FW/NAT/Router with two interfaces. My local phones are situated behind the NAT and connects to the outer interface of the */FW/NAT/Router. * is then connected to my SIP providers (since I'm only using the SIP-part of *, PSTN connection through my SIP-provider). Works fine! rgds, /staffan kerker sweden -Ursprungligt meddelande- Från: Steven M. Sokol [mailto:[EMAIL PROTECTED] Skickat: den 22 oktober 2003 17:22 Till: [EMAIL PROTECTED] Ämne: [Asterisk-Users] Running Asterisk and NAT on the same box? Has anyone tried installing * on a box with two eth interfaces which is acting as a NAT box? I have only one IP at this point and I would like to get * working without all of the NAT issues. My idea is to run * on my gateway (which is also running the firewall and masquerade services). All of my UAs (Grandstream + Xten X-LITE + gnophone) will be inside the NAT screen, and will connect to the * using its PUBLIC (outside) address. Does this sound reasonable? Thanks, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Placing SIP calls to other SIP domains?
Hi! Does * do DNS-lookups when outgoing calls are placed to a different SIP domain? Can I call from to ? Can * work as a regular SIP proxy in that aspect? Can * handle SIP URI:s that are complete SIP URI:s (sip:[EMAIL PROTECTED]) instead of numbers only? Or should I run a SIP proxy on a different machine to handle pure SIP requests and let * register an account? rgds, /staffan kerker sweden ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
