Re: [asterisk-dev] asterisk broke and I'm getting fired from
Start Asterisk as asterisk -cvvv jazzy singh wrote: I did that but it still is dyingjust to let you guys know I'm using lumenvox as well, when I restart asterisk it restarts and then it just dies right away without any error message. Please help Thanks Date: Thu, 26 Jul 2007 14:20:59 -0400 From: harish kasiviswanathan [EMAIL PROTECTED] Subject: Re: [asterisk-dev] asterisk broke and I'm getting fired from my job:( To: Asterisk Developers Mailing List asterisk-dev@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-13 Looks like you are using the old version of zaptel driver and new version of asterisk. Upgrade to zaptel-1.4.4 or newer and this error should go away. harish On 7/26/07, jazzy singh [EMAIL PROTECTED] wrote: I was running a successfull asterisk box but i don't know what happened for some reason asterisk shuts down as soon as I start it. It won't give me any error or anything, but last nite when I was forking through error log I did see this though Jul 25 23:02:55 NOTICE[8329] chan_zap.c: Got event 18 (Ring Begin)... Jul 25 23:02:55 WARNING[6651] channel.c: Avoided initial deadlock for '0x8a94f88', 10 retries! Jul 25 23:17:39 NOTICE[9303] chan_zap.c: Got event 18 (Ring Begin)... and then i tried to recompile asterisk and it gave me this codec_zap.c: In function ?find_transcoders?: codec_zap.c:833: error: variable ?info? has initializer but incomplete type codec_zap.c:833: warning: excess elements in struct initializer codec_zap.c:833: warning: (near initialization for ?info?) codec_zap.c:833: error: storage size of ?info? isn?t known codec_zap.c:838: error: ?ZT_TCOP_GETINFO? undeclared (first use in this function ) codec_zap.c:848: error: ?ZT_TRANSCODE_OP? undeclared (first use in this function ) codec_zap.c:833: warning: unused variable ?info? make[1]: *** [codec_zap.o] Error 1 make[1]: Leaving directory `/downloads/asterisk-1.2.20/codecs' make: *** [subdirs] Error 1 I was able to compile and install the same source code before. Please someone help me if I don't get this to work today I might get fired :(.. please help. thanks in advance -- Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Dial() Staggering
William Moore wrote:
Dial() already supports dialing multiple lines at once and connecting the
first answered line to the user. I want that exact same functionality but
would like to be able to space out (or stagger) the dialing of each line.
An example might be I want to dial 5 phone numbers with a wait time of 30
seconds. But I want it to try the first number for 2 seconds before
starting on the next number. Then at that point it would be ringing 2
numbers (with 2 seconds between when each was dialed).
This can be accomplished in the dialplan by calling the first number
for x seconds (using Dial's timeout option) and when it doesn't
answer, use a ring group to call several people at once. I admit it's
not *EXACTLY* what you're looking for, but it is close enough to be
usable.
The line wraps make this suck. Anyway, add a Wait(10) as the first
priority of the main-1 main-2, etc to delay that.
This assume each line appearance is a separate SIP user ID.
exten = 3800,1,Set(CFU_DEST=${DB(${EXTEN}/CFU)})
exten = 3800,2,GotoIf($[${LEN(${CFU_DEST})} =
0]?3:toll-access,${CFU_DEST},1)
exten =
3800,3,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED],24)
exten = 3800,4,Voicemail([EMAIL PROTECTED],u)
exten = main-1,1,Dial(SIP/0004f20621b3-b)
exten = main-1,2,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?3:10)
exten = main-1,3,Dial(SIP/0004f20621b3-c)
exten = main-1,4,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?5:10)
exten = main-1,5,Dial(SIP/0004f20621b3-d)
exten = main-1,6,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?7:10)
exten = main-1,7,Dial(SIP/0004f20621b3-e)
exten = main-1,8,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?9:10)
exten = main-1,9,Dial(SIP/0004f20621b3-f)
exten = main-1,10,Hangup
exten = main-2,1,Dial(SIP/0004f2062301-b)
exten = main-2,2,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?3:10)
exten = main-2,3,Dial(SIP/0004f2062301-c)
exten = main-2,4,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?5:10)
exten = main-2,5,Dial(SIP/0004f2062301-d)
exten = main-2,6,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?7:10)
exten = main-2,7,Dial(SIP/0004f2062301-e)
exten = main-2,8,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?9:10)
exten = main-2,9,Dial(SIP/0004f2062301-f)
exten = main-2,10,Hangup
exten = main-3,1,Dial(SIP/0004f2062135-b)
exten = main-3,2,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?3:10)
exten = main-3,3,Dial(SIP/0004f2062135-c)
exten = main-3,4,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?5:10)
exten = main-3,5,Dial(SIP/0004f2062135-d)
exten = main-3,6,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?7:10)
exten = main-3,7,Dial(SIP/0004f2062135-e)
exten = main-3,8,GotoIf($[${DIALSTATUS} = BUSY | ${DIALSTATUS} =
CHANUNAVAIL | ${DIALSTATUS} = CONGESTION]?9:10)
exten = main-3,9,Dial(SIP/0004f2062135-f)
exten = main-3,10,Hangup
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Re: [asterisk-dev] Docs converted to TeX?
Russell Bryant wrote: Eric ManxPower Wieling wrote: I discovered today that in -trunk all the docs have been converted to TeX format. Are these docs the only ones that will be in the release tarball? If so, I think it is a bad idea. We do not want to make people install yet another dependency to be able to read the basic Asterisk documentation. By the time Asterisk 1.6 comes around, I plan on editing the Asterisk release script to automatically include some user friendly format of the documentation in the tarball. I am thinking HTML would probably be best. Thank you. Having a plaintext output would be nice as well. I don't run a GUI on my Asterisk boxes. Having to download the HTML docs to a machine with a GUI and a web browser would also be a hassle just to look up 1 page of docs. Even with things like URPMI/YUM/APT-GET, getting TeX installed would be a significant hassle. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] chan_cellphone - Mantis issue 8919
Andrew Kohlsmith wrote: On Thursday 10 May 2007 2:26 pm, Olle E Johansson wrote: chan_pan :-) PAN/nokia PAN/peter PAN = Personal Area Network Eww... this isn't using the PAN profile at all, so I don't think that'd be right... chan_HFP or HSP would be my guess if you wanted to go that route... but yeah chan_bluetooth should really be what this should be called, since it'll handle damn near everything. Name it something totally off the wall like chan_cthulhu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Adding Octastic Soft-Echo to external SIP adapters
syd wonder wrote: Hi all. I'm trying to hook into a chan_zap.c function that enables echo cancellation. I think I've identified the spots to hook into but am wondering if someone can answer a few questions or provide some guidance. 1 - I'm using an external device (Sipura SPA3000 FXS/FXO) configured as a SIP trunk. I'm hoping that along with the Chan_Sip.C file, that the Chan_Zap.c file is used to setup some of the functionality, namely - whether echo cancellation should be enabled 2 - With item 1 being true, the reason why echo cancelling is not being enabled is because the external device is being recognized as a digital channel. Anyone have a suggestion as to whether these assumptions are true? If not, any suggestions how I can add the echo cancellation functions to non-zap devices. This is something a bit more interesting now that Octastic has released their SoftEcho software. Thanks.. Syd Generally, once the audio is converted to VoIP audio latency is far too high for echo canceling to work. Echo needs to be canceled BEFORE the call is converted to VoIP. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Adding Octastic Soft-Echo to external SIP adapters
Tzafrir Cohen wrote: On Mon, May 07, 2007 at 02:45:48AM -0500, Eric ManxPower Wieling wrote: syd wonder wrote: Hi all. I'm trying to hook into a chan_zap.c function that enables echo cancellation. I think I've identified the spots to hook into but am wondering if someone can answer a few questions or provide some guidance. 1 - I'm using an external device (Sipura SPA3000 FXS/FXO) configured as a SIP trunk. I'm hoping that along with the Chan_Sip.C file, that the Chan_Zap.c file is used to setup some of the functionality, namely - whether echo cancellation should be enabled 2 - With item 1 being true, the reason why echo cancelling is not being enabled is because the external device is being recognized as a digital channel. Anyone have a suggestion as to whether these assumptions are true? If not, any suggestions how I can add the echo cancellation functions to non-zap devices. This is something a bit more interesting now that Octastic has released their SoftEcho software. Thanks.. Syd Generally, once the audio is converted to VoIP audio latency is far too high for echo canceling to work. Echo needs to be canceled BEFORE the call is converted to VoIP. So a PRI card with echo cancelling won't help local SIP phones, right? It will if the call is going out the PRI. 8-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Adding Octastic Soft-Echo to external SIP adapters
Klaus Darilion wrote: Vazir wrote: How than my CISCO AS5350 gets H323 VOIP and echo-cancels so exelently :)) If it is a plain H323-H323 call then there should not be any echo at all. If you have a PSTN-VoIP call and the PSTN leg uses analog lines then the PSTN side will create an echo which will be cancelled by the PSTN side of the gateway. In my experience, most echo comes from the FAR end analog loop, not your local analog loop. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] pickup call groups
Just how many pickup groups do you need? If you are assigning one pickup group for each extension then your design is wrong. Pavel Jezek wrote: if this limitation is really true, it is challenge for some rework, because with 64 pickup groups limit, it's usefull only for small companies :-( Philipp Kempgen wrote: Dov Bigio wrote: Is there any possibily of having more than 0-63 pickup/callgroups Not unless you invent something on your own. They are stored as a long long int which has 64 bits so this is not easily extendable. You could do some sort of checking for a user's permissions before doing the Pickup(). Regards, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] callerid on redirected calls
Roy Sigurd Karlsbakk wrote:
hi all
I've been fighting this for a while, and telco tells me i'm doing it
wrong and all, although I'm doing it after the book (or so I beleive).
To divert a call, I do as follows
exten = s,n,Set(CALLERID(rdnis)=${CALLERID(number)})
exten = s,n,Set(CALLERID(number)=${EXTEN})
exten = s,n,Set(CALLERID(ani)=${EXTEN})
exten = s,n,Dial(${DIVERT_TO_NUMEBER})
You are setting the Caller*ID Number to s. I doubt your telco would
like that.
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Re: [asterisk-dev] SIP Multiple endpoints with same id
No it is not possible. Alexei Volkov wrote: Is it possible (in theory) to make asterisk server multiple sip endponts configured with same sip credentials. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Clearing pick-up groups on Zap/ channels
Why not allow group= or make group 0 mean no group. What I do is set channels that I don't want in a group to be group 0 and never use g0 anywhere Nic Bellamy wrote: Olle E Johansson wrote: 5 nov 2006 kl. 16.43 skrev Gil Kloepfer: I've discovered that in configuration files such as the one for chan_zap (where all the options cascade down rather than being specific for each category, there is no way to indicate that channels have no pick-up group after the first group has been set. For example (simplified), in zapata.conf: [channels] ; These two phones are in call group/pick-up group #1 callerid=Green Phone(256) 428-6121 callgroup=1 pickupgroup=1 channel = 1 callerid=Black Phone(256) 428-6122 channel = 2 ; These three channels should not be in ANY pick-up groups, but there ; is no way to clear the previous settings callerid=CallerID Phone (256) 428-6123 channel = 3 callerid=CallerID Phone (256) 704-4666 channel = 4 callerid=CallerID Phone (630) 372-1564 channel = 5 In the case of channels 3, 4, and 5, there is no valid way to clear out the current callgroup / pickupgroup without encountering an error (you could say pickupgroup=none, but that actually throws an error from ast_get_group(), although it happens to work). I'd like to propose making a change to ast_get_group() to allow the option none that returns a ast_group_t containing no groups set, basically zero. So you could say 'callgroup=none' for example. Note that callgroup=0 would not do what I am suggesting because 0 is a valid group number. The group/callgroup/pickupgroup is actually a bit mask (bits numbered 0 to 63). If this seems reasonable, please indicate so and I will submit a patch for both 1.2 and -trunk (they are essentially the same patch). Sounds very reasonable. Whether we can see this as a bug or a new feature is up to Russell to decide. If it's a bug fix, which I think, we need patches for the 1.2 and 1.4 branches plus trunk. Please open an issue in the bug tracker, upload patches and we'll discuss there. I've been trying to think of an easy, minimal-change way out of the zapata.conf inheritence problem (since it's not just pickupgroups that have this behaviour, it's just worse with that since you can't reset it at present). What about something simple like a resetdefaults item that will restore all zapata.conf settings to hardcoded defaults, and clear out the pickupgroup/callgroup stuff? Ie. pickupgroup=1 callgroup=1 channel = 1-3 resetdefaults=yes ; likely need the =yes since it's key=value based otherconfig=123 channel = 4 This wouldn't break any existing configs out there, since it'd only happen if you explicitly used resetdefaults. Thoughts? Or shall I just whip up a patch? :-) Regards, Nic. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Re: How to busy out PRI channels?
Tony Mountifield wrote: In article [EMAIL PROTECTED], Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Alistair Cunningham wrote: When I was an engineer working mostly with IBM Websphere Voice Response, and we wanted to take a machine down, we would quiesce the trunk(s) as above, then wait until only 1 or 2 calls were left, then cut off these remaining calls. We felt this was the best compromise between cutting off as few callers as possible, and minimising the time the system was unavailable to new callers. A facility to do this in Asterisk would be most welcome. stop gracefully does something similar. But does it stop new calls coming in while it's waiting for the existing calls to finish? I suspect not. Yes. stop gracefully does not allow additional calls and does not terminate existing calls. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] bug or feature (use From: instead of Digest username to match INVITE) ?
Tzafrir Cohen wrote: On Mon, Oct 16, 2006 at 06:43:53AM -0500, Rich Adamson wrote: What I'm worried about is trying to fix it in the current code, since it will change quite a few things that are needed today, and break backwards compatibility. I've tried, but failed, in chan_sip2 :-) and that was when I was beginning to start thinking of chan_sip3, that will break backwards compatibility :-) But, didn't we run across that same compatibiliity issue with iax and iax2, letting both peacefully exist until some predetermined date when iax was remove? Wouldn't the same approach work with sip3? IAX2 was technically a different protocol. That is: it used a different port number. However chan_sip1 and chan_sip3 are both implementations of the same protocol, and expected to listen on the same port 5060. So you could run both on the same system, if one is bound to a different port. But if a remote system calls in and asks to connect through SIP, you have to choose, as it will typically just expect it on the well known SIP port. Sounds more like Voicemail and Voicemail2 situation. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] libwrap
Peter Beckman wrote: Hey folks -- Noticed that Asterisk doesn't use libwrap. Any reason? Could it be added? It would be handy. Does libwrap support UDP? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] bug in echo cancel at 256 taps
Sounds like ECFO. Echo Canceler Freak Out, this happens when the rxgain is too high and the echo canceler freaks out. Some users describe it as screeching, feedback, static, or other useless terms. If users report static on a system where there cannot be static (all digital, PRI, SIP, etc), you might be experiencing ECFO. John Lange wrote: Yesterday and this morning we experienced a serious bug in the echo cancellation. While trying to tweak out some echo on a new asterisk install we set echocancel=256. From then on, calls would usually start out ok (but still with some echo) then there would be what users describe as a 'click' followed by a huge blast of echo making the call impossible to continue. At that point it seems as though the echo is actually being inserted into the call rather than removed. asterisk-1.2.10 libpri-1.2.3 zaptel-1.2.7 with default ECHO_CAN_KB1 Card is a Sangoma A101 using wanpipe-beta7-2.3.4.tgz . Just wanted to run it by the list before entering a bug report to make sure its appropriate and that this isn't actually a wanpipe bug. Is there any other sort of other debugging or analysis that would be helpful? Regards, John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Asterisk Core Dumps
/path/to/src/asterisk/doc/README.backtrace Kohler, Jeffrey wrote: I was able to eventually figure it out. For anyone as linux unsavvy as myself: - # gdb /usr/sbin/asterisk /tmp/core.21713 - # bt Sorry for the stupid question From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kohler, Jeffrey Sent: Friday, July 21, 2006 8:21 AM To: Asterisk Developers Mailing List Subject: [asterisk-dev] Asterisk Core Dumps I've been experiencing some Asterisk crashes and was trying to look at the core dumps to figure out what is going wrong. Unfortunately I'm not really even sure where to start. (I'm a Windows developer by trade so please excuse my ignorance) Here's what I've done so far: - run Asterisk with the safe_asterisk script - when Asterisk crashed it produced a dump file - core.21713 - From a command line, I ran gdb - # core-file /tmp/core.21713 - # bt - I did get a stack trace, however, it simply lists a series of addresses. How can I get symbolic information to make sense of this? Is there a better method of examining core files? -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Proposal to seperate qualify keep alive
SIPura has that feature. Kevin P. Fleming wrote: - John Lange [EMAIL PROTECTED] wrote: The question then becomes, if most client devices support keep-alive is there still a purpose to having it on the server side as well? How many client devices support keep-alive? I know Linksys products do but I haven't looked into others yet. I've never seen the devices I've worked with supply that functionality, although I've not looked specifically for it. I think it will always be necessary from the server end. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Making some changes to chan_sip and wouldlikesomefeedback
Alejandro Kauffmann wrote: Eric ManxPower Wieling wrote: My only comment is that using extension to refer to an entry in the current sip.conf is insane. SIP devices are not extensions. They are devices which may or may not have more than one entry in sip.conf (one for each line/call appearance). Perhaps sip_account.conf would be a better term. Agreed extension was a poor choice. We are actually thinking in terms of devices, but had not considered multiple appearance devices. Perhaps account.conf (since we are placing them in a sip directory) or sip_account.conf (as suggested) would be more accurate. We use the MAC address as the SIP User ID. For example the Polycom IP 50x have 3 call appearances. We register each line as a separate SIP user ID. First line MAC-a, second line MAC-b, etc. This helps to prevent anyone from thinking a device is an extension. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] bug or bad chan_sip.c
Your sip.conf needs to accept calls from anyone. [EMAIL PROTECTED] wrote: Hello, Anybody could explain me why asterisk spend time to send back to proxy or sip agent authentication messages 407 nobody can call me from other domains. can we disable authentication for none peers or users Asterisk ask authentication 407 for sip:[EMAIL PROTECTED] or sip:[EMAIL PROTECTED] [sip] include = info include = support exten = info,1,Answer() exten = info,2,Dial(Sip/84,10) exten = info,3,Dial(Sip/85,10) exten = info,4,Hangup exten = support,1,Answer() exten = support,2,Queue(support|t||) exten = support,3,Hangup ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] asterisk developpers
Asterisk assumes all incoming calls will be authenticated. You want all incoming calls to not require authentication. This is not a common usage of Asterisk. There is an option called insecure=very This option is shown in the /path/to/src/asterisk/configs/sip.conf.sample I assume the Wiki would also have information about this. [EMAIL PROTECTED] wrote: I waste time to configure a simple feature . you can read my posts. Why asterisk is not able to manage authorisation, authentication in easier way ? HarryI waste time to configure a simple feature . you can read my posts. Why asterisk is not able to manage authorisation, authentication in easier way ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] Status of another channel from AGI
Steven Critchfield wrote: On Fri, 2006-03-03 at 11:20 -0500, Paul wrote: Steven wrote: snip Performance and security is not special to any interface of asterisk. Performance isn't usually a quality of AGI that is taken really seriously. Look at the many applications Tilghman has written so that one doesn't need to spawn out to AGI to accomplish certain tasks. He has mentioned before that he is trying to eliminate most needs for AGI and help keep the development in the dialplan. If I were to take an existing dialplan and simply change all existing AGI invocations to use macros would it affect performance other than startup and reload when the macros are parsed? That would allow me to replace AGI with applications by simply changing a macro. Startup is where my company is seeing the biggest hit. Granted our app is a pretty large perl app, but it takes more than a second and sometimes as much as two seconds to start. We have had to mask startup by providing ringing after answer. Isn't this what FastAGI was designed for? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
