RE: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 w/o j itterbuffer enabled?
-Original Message- From: Benjamin on Asterisk Mailing Lists [mailto:[EMAIL PROTECTED] Sent: September 7, 2004 10:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2 w/o jitterbuffer enabled? On Tue, 7 Sep 2004 16:26:24 -0700, Kris Boutilier [EMAIL PROTECTED] wrote: I'm having a problem with intersite calls over IAX2 being abruptly terminated. Nothing odd shows in any of the logs for Asterisk or the host. The only think I can think it might be is a lag-spike on the site to site connection. When does the cut off occurr? Is it always after about 8-10 seconds? If so, you may have a problem with IAX transfer. You can verify this by using notransfer=yes. Having notransfer=no is beginning to sound like it's the culprit, but I don't understand why. I don't see any related issues in bugs.digium.com nor is there any intervening network equipment that would interfere with the transit of packets between hosts (such as a NAT firewall). The only thing that springs to mind, if it's all UDP driven, is a lack of retry handler for the UDP handoff acknowledgement? I'm averaging about a 0.5% collision rate on this network (half-duplex 10Base-T)... Kris Boutilier Information Systems Coordinator Sunshine Coast Regional District ___ 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] Compiling on Mac OS X (10.3.5)
On Wed, 8 Sep 2004 01:04:53 +0100, asterisk-users [EMAIL PROTECTED] wrote: I have successfully used the packaged version of * on the Mac for some time, but decided that I would recompile one of the more recent builds so that my PC and Mac were in sync. As suggested, I installed the XCode tools, updated bison and downloaded the latest version of *. Unfortunately, when compiling there are lots of errors, many of them relating to the non-existent /usr/src/linux directory. Please let me know which version of Asterisk you were trying to build and also send me the session transcript of your compile and build trial and I'll take a look at it. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] DTMF Caller ID w/o polarity inversion
Renato Mintz wrote: Hi Folks, I've been looking around and found some references of some Caller ID patches (Mantis bug#9) for X100P and TDM400 for Netherlands, Sweden and UK. It's been quite hard to understand what has finally been incorporated to the distribution (if anything) or which patches must be applied in witch snapshot of the repository. I've tried some different approaches but nothing worked and my question finally is: Is there any implementation for X100P or TDM400 that supports DTMF caller ID WITHOUT the need of polarity inversion before the DTMF spill? Is anyone working on this? This is the way it works in Brazil and some other coutries... Thanks a lot, Renato Renato, Bug id=9 in the bugtracker is not currently inserted into CVS but can be applied to it. The additional patch for X100P that I put there (srathje) is a quick (ugly) hack for the Danish CID system that uses the DTMF decoder made by egnarf. In Denmark we have no warning (officially) before the CID is received but by monitoring the line I found that a short burst or signal is received just before the CID so I modified the wcfxo.c code to look for this. There is a different approach in bug=1719 for monitoring UK BT CID as it seems to be the same problem, no warning before the CID (V23 FSK) is received. The method used here is a history buffer used to capture the CID and decode when the first ring is detected. I guess you will have to investigate both solutions and use what you can to make a viable solution. If you are successfull in doing so please share with others via the bugtracker.. :-) Regards Soren Rathje ___ 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] Problems with length of voicemail
I had that problem when I was running asterisk on my linux box before it went down so you aren't the only one having that problem - Original Message - From: Marty Mastera To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, September 07, 2004 10:50 PM Subject: RE: [Asterisk-Users] Problems with length of voicemail I wonder if anyone else's Asterisk box drops the connection to voicemail after 30 secs even when the maxmessage parameter is set to 180 (3 mins). Here is the general section of my voicemail: Roger, There has been very recent discussion regarding this topic exactly...specifically when using BroadVoice as a sip provider. Calls toyour BroadVoice DID that end up in VM terminate after 30 seconds The current theory is that during VM recording, * doesn't send any audio packets back to BroadVoice...after 30 seconds BroadVoice thinks that the connection has been lost and terminates the call...(I'm paraphrasing the thread that recently appeared on this topic, forgive me if this isn't completely accurate) Assuming that this is correct, you could be using BroadVoice, or another provider who disconnects after not receiving audio for some period of time... Hope that helps, Marty ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users My Inbox is protected by SPAMfighter983 spam mails have been blocked so far.Download free SPAMfighter today!___ 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] astwind has any one got this thing to work?
hello I am fitteling with the astwind-installer-0.1.1.exe asterisk for windows and am having trouble getting the thing to connect to the meers to download the updates and stuff. I looked at the wiki and set up networking and stuff with no success, has any one got this thing to work successfully? my windows box is the faster of the 2 machines and my main linux box is down at the moment. I am running a netgear rp614 router behind nat if this helps but I have tried and tried and tried to get this sucker up with no luck any help would be greatly greatly appreciated. thanks hank My Inbox is protected by SPAMfighter983 spam mails have been blocked so far.Download free SPAMfighter today! ___ 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] Answer confirmation on non-Zap channels?
I was looking at the sample "follow me" config (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) which uses a dial modifier 'c' to enable Answer confirmation - "If the letter c follows, then "Answer Confirmation" is requested, in which the call is not considered answered until the called user presses #. " (http://www.voip-info.org/wiki-Asterisk+ZAP+channels). I would like to get this feature working on non-Zap channels such as IAX2 or SIP. My need is exactly as the "follow me" example describes: Ring my desk,if no answer ring my cell andif no answer revert back to * for VM...Answer confirmation seems to achieve my goal of not letting calls end up in my cell phoneVM by requiring a '#' keypress from the called user (me on my cell) to consider the call answeredMy other thought is to time how long it takes for an unanswered call to end up in my cell VM and limit * to dialing the cell for n-3 seconds or something similar to avoid ever hitting the cell VM...problem is, if I'm out of coverage or if the phone is turned off, the cell will answer the call immediately... If this is not implemented in other channels, can anyone recommend either an alternative solution or a starting point for implementing it in the code of other channels? Thanks, Marty Mastera M3 Resources [EMAIL PROTECTED] Phone: 303.680.1283 x200 FAX: 303.680.1283 IAXTel: 700.206.7507 FWD: 484162 ___ 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] Answer confirmation on non-Zap channels?
Hi Marty
I think this is a valuable add-on to my feature request to play audio
until # is pressed - see
http://bugs.digium.com/bug_view_page.php?bug_id=0002356
At the moment just hearing silence is not very helpful to users.
Perhaps we can extend this feature to
a) provide this support on multiple channels. I think this may mean
moving it out of chan_zap.c and putting it into app_dial.c to make it
more generic.
b) Provide the ability to playback multiple files or messages to the
called party whilst waiting for answer. e.g.
Background(you have a call from)
SayDigits(${CALLERIDNUM}}
Background(press # to accept, * to deflect to voicemail, or hang up to
ignore...}
and only send answerback to the called party when the call is finally
accepted by pressing the # key.
Anyone else got any suggestions - or inclination to code this?!
Thanks
Tim
Marty Mastera wrote:
I was looking at the sample follow me config
(http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) which uses a
dial modifier 'c' to enable Answer confirmation - If the letter /c/
follows, then Answer Confirmation is requested, in which the call is
not considered answered until the /called/ user presses *#*.
(http://www.voip-info.org/wiki-Asterisk+ZAP+channels). I would like
to get this feature working on non-Zap channels such as IAX2 or SIP.
My need is exactly as the follow me example describes: Ring my
desk, if no answer ring my cell and if no answer revert back to * for
VM...Answer confirmation seems to achieve my goal of not letting calls
end up in my cell phone VM by requiring a '#' keypress from the called
user (me on my cell) to consider the call answeredMy other thought
is to time how long it takes for an unanswered call to end up in my
cell VM and limit * to dialing the cell for n-3 seconds or something
similar to avoid ever hitting the cell VM...problem is, if I'm out of
coverage or if the phone is turned off, the cell will answer the call
immediately...
If this is not implemented in other channels, can anyone recommend
either an alternative solution or a starting point for implementing it
in the code of other channels?
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RE: [Asterisk-Users] Answer confirmation on non-Zap channels?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tim Robinson
Sent: Wednesday, September 08, 2004 1:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Answer confirmation on non-Zap channels?
Hi Marty
I think this is a valuable add-on to my feature request to
play audio until # is pressed - see
http://bugs.digium.com/bug_view_page.php?bug_id=0002356
At the moment just hearing silence is not very helpful to users.
Perhaps we can extend this feature to
a) provide this support on multiple channels. I think this
may mean moving it out of chan_zap.c and putting it into
app_dial.c to make it more generic.
b) Provide the ability to playback multiple files or messages
to the called party whilst waiting for answer. e.g.
Background(you have a call from)
SayDigits(${CALLERIDNUM}}
Background(press # to accept, * to deflect to voicemail, or
hang up to ignore...}
and only send answerback to the called party when the call is
finally accepted by pressing the # key.
Anyone else got any suggestions - or inclination to code this?!
Thanks
Tim
Hey Tim,
I wholeheartedly endorse the idea of making this more generic and not
channel specific...as to your ideas, I would be happy with having to
press '#' to indicate acceptance of the call, even if there is only
silence on the other end. On the other hand, I like your ideas of
announcing the call, verbalizing the callerid digits and presenting a
menu to answer or deflect, etc...that would be icing and very cool...
I'm happy to contribute on this topic where possible...I'm not a coder
but I'm willing to take the effort to learn whatever will make me useful
towards this goal...
Marty
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Re: [Asterisk-Users] DTMF Caller ID w/o polarity inversion
In France we just have a very short ring before the CID spill.
The CID spill is encoded in V23 instead of Bell 202
If you want the X100P to decode CID try this in fskmodem.c :
#define FLIST {1400,1800,1300,2100}
and maybe (i don't remember !) callerid.c :
#define CALLERID_SPACE 2100.0 // CCITT V23
#define CALLERID_MARK 1300.0
If you want the TDM400 to send the cid see :
http://bugs.digium.com/bug_view_page.php?bug_id=600
- Original Message -
From: Renato Mintz [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, September 08, 2004 3:00 AM
Subject: [Asterisk-Users] DTMF Caller ID w/o polarity inversion
Hi Folks,
I've been looking around and found some references of some Caller ID
patches (Mantis bug#9) for X100P and TDM400 for Netherlands, Sweden
and UK. It's been quite hard to understand what has finally been
incorporated to the distribution (if anything) or which patches must
be applied in witch snapshot of the repository.
I've tried some different approaches but nothing worked and my
question finally is:
Is there any implementation for X100P or TDM400 that supports DTMF
caller ID WITHOUT the need of polarity inversion before the DTMF
spill? Is anyone working on this? This is the way it works in Brazil
and some other coutries...
Thanks a lot,
Renato
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[Asterisk-Users] 'connecting' voip-numbers to our Asterisk
Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf * [ip-incoming] exten = ,1,Dial(106,20,r) * /etc/asterisk/iax.conf * register = :[EMAIL PROTECTED] * This should be all I need to let incoming calls on ring on extension 106, right? Regards, Evert ___ 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] Newbie: Only allow authenticated users to call
I made the observation that I'm able to make a call with my SIP client (kphone) even when I'm not registered/authenticated. Of course, when I'm not registered at asterisk, people can't call me, but it's still a huge security hole, that unregistered Clients can make calls. Is there a way to tell asterisk to only allow registered clients making calls? I know about the Anti Ex Girlfriend function, but this is not what I want. Regrads, Henry ___ 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] 'connecting' voip-numbers to our Asterisk
On Wed, 08 Sep 2004 10:08:00 +0200, Evert Meulie [EMAIL PROTECTED] wrote: I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf * [ip-incoming] exten = ,1,Dial(106,20,r) * /etc/asterisk/iax.conf * register = :[EMAIL PROTECTED] * This should be all I need to let incoming calls on ring on extension 106, right? No. First of all, let me ask you this... Are you sure that this provider supports IAX? I am asking because the Cisco 7960 doesn't do IAX, so you wouldn't have been using IAX when connecting directly. Second, if your provider does support IAX, then you will also need to set up a peer for incoming connections and send the calls to your incoming context, like so ... [iaxprovider] type=user username=888 secret=blah host=iax.provider.com qualify=yes disallow=all allow=whatever-codec-they-support context=incoming-from-iaxprovider this may or may not work depending on how your provider will try to connect to you. For example, FWD will always come in as user iaxfwd, so if you don't define your inbound peer as [iaxfwd] it won't work. Also, some providers use passwords, others use RSA keys. but assuming that the above matches the way in which your provider expects to connect to you, then you will still need an incoming context in extensions.conf named the same way as whatever comes after the context= setting. Even that may not be enough depending on how your proider presents the call to you. They may come in using your username or number, but they may as well use an account code or simply s. You will have to check out the sample configuration or whatever other documentation they provide. The chance is that somebody on this list is using the same provider, so you may tell us what provider you are using and somebody may then share their configuration with you. Also, the Wiki may have a sample configuration for the provider you are using. I always use the IAX debug command on the console to find out how an IAX peer comes in. Simply enter the command iax2 debug on the Asterisk console, then make a test call and see what the debug output says. It's pretty self explanatory. Use the command iax no debug to turn debugging off again. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] Music on hold problem
Have you tried running mpg123 from the command line? I found that it was failing to load the mp3 cos it couldnt open /dev/dsp. If this is the case, then I found the following worked, although it is a little OTT I guess... If you have a soundcard in the machine try insmod'ing a driver for it and then as root, run 'rm -f /dev/dsp*' and then '/dev/MAKEDEV audio'. If you dont have a soundcard then try just loading the sound module, that might just be enough. Then try again. Ensure you stop and start asterisk again to ensure MOH reinitialises properly. jd -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Karim Mardhani Sent: 08 September 2004 05:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Music on hold problem Hi All: I am having problems with getting music on hold to work. I get following error Executing SetMusicOnHold(SIP/2000-2ddf, default) in new stack -- Executing Answer(SIP/2000-2ddf, ) in new stack -- Executing WaitMusicOnHold(SIP/2000-2ddf, 30) in new stack Sep 7 15:50:47 WARNING[1190886320]: res_musiconhold.c:329 moh1_exec: Unable to start music on hold (class '30') on channel SIP/2000-2ddf == Spawn extension (from-sip, 4999, 3) exited non-zero on 'SIP/2000-2ddf' asterisk*CLI I have made sure that I have mpg123 in /usr/bin. The version of mpg123 I have is 0.59r. Following is my extension.conf: exten = 4999,1,SetMusicOnHold(default) exten = 4999,2,Answer exten = 4999,3,WaitMusicOnHold(30) exten = 4999,4,Hangup When I change WaitMusicOnHold with MP3Player(/var/lib/asterisk/mohmp3/sample-hold.mp3) I can hear the sample-hold file when I dail extension 4999. Do both MP3Player and MOH use mpg123? Any help in this regard would greatly be appreciated. Thanks Karim ___ 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 This message has been scanned for viruses by MailController - www.MailController.altohiway.com --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.745 / Virus Database: 497 - Release Date: 27/08/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.745 / Virus Database: 497 - Release Date: 27/08/2004 ___ 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] Newbie: Only allow authenticated users to call
On Wed, 8 Sep 2004 10:31:44 +0200, Henry Jensen [EMAIL PROTECTED] wrote: I made the observation that I'm able to make a call with my SIP client (kphone) even when I'm not registered/authenticated. Of course, when I'm not registered at asterisk, people can't call me, but it's still a huge security hole, that unregistered Clients can make calls. Make sure you don't include your default context anywhere you don't want unregistered callers to have access to. This also means you shouldn't have any extensions in your default context that you don't want unregistered callers to have access to. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] astcc dont write to the table cdrs or cards
Variables DIALSTATUS: added to CVS head in june/july 2004! What is your CVS version? Areski On Wed, 2004-09-08 at 03:44, Doug Harris wrote: Hi, I have set-up astcc with outgoing sip channel. Call processing works fine but after the call tables, CDR and Cards does not get updated. At the beginning it goes to the database and fetch card details and correctly provides the card balance etc. Also it indeed write the inuse field (so writing and reading from database works fine). I've inserted a break point as such in the code; $dialstatus = $AGI-get_variable(DIALSTATUS); print STDERR dial status $dialstatus\n; It seems like dialstatus is not returned (which prints nothing). So obviously later part of the agi does not go through database updating portion (which only happens if dialstatus = Answerd). I am using deadagi to call the astcc.agi script as explained. Can someone explain why this happens ? Cheers dh __ ___ 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] X-Lite Meetme problem
HI! Have a weird problem with X-lite Meetme. When X-Lite user are join to conference room NOT first one, than X-Lite user do not hear anything. This problem gone when X-Lite user get into conference room first (when nobody there). sip.conf [104] context=VoIP-only type=friend username=104 secret=test host=dynamic dtmfmode=rfc2833 mailbox=104 canreinvite=no disallow=all allow=ulaw ;allow=alaw ;allow=gsm On * console have such messages: when X-Lite using ULAW: Sep 8 09:47:17 WARNING[1233853360]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/4) on x-lite ALAW Sep 8 09:49:12 WARNING[1235344304]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/8) on x-lite GSM Sep 8 09:51:50 WARNING[1233804208]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 2 (read/write = 64/2) Please advice. -- Best regards Vlad ___ 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] Newbie: Only allow authenticated users to call
I'm wondering if you are confusing two ideas. It has to be possible for anyone to be able to call you just like they can on an ordinary POTS line. Registration is for those who need to appear in some sense internal to the PBX. Using dialplan contexts you can offer very different functionality to callers who are registered versus those who are just calling. For example, you might assign all registered users to a context call internal and provide access to all the dial plans. You might set the context of all non-registered callers to an external dialplan context. The internal context might provide access to all the telephony services an internal user might expect (eg dial 7 to get to voicemail automatically). The external context might direct a caller to the operator or to a voice prompt. Optionally, you might provide an extension for voicemail so that external employees calling from home or a client site can get to their messages. Clearly the caller will need to be prompted for a voice mail box and password but that's covered by the voicemail system. Bill Seddon Lyquidity Solutions -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Jensen Sent: September 08, 2004 9:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Newbie: Only allow authenticated users to call I made the observation that I'm able to make a call with my SIP client (kphone) even when I'm not registered/authenticated. Of course, when I'm not registered at asterisk, people can't call me, but it's still a huge security hole, that unregistered Clients can make calls. Is there a way to tell asterisk to only allow registered clients making calls? I know about the Anti Ex Girlfriend function, but this is not what I want. Regrads, Henry ___ 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] re: asterisk, SER and autocreatepeer
Hi all, quick question...i am using autocreatepeer to get asterisk to work with SER without having to specify each UA in sip.conf and in ser separately. 2 questions: 1. obviously this is not very secure because anyone can bypass the SER and register themselves as a peer with the asterisk. assuming i block incoming requests on the port asterisk is running SIP on (excluding requests from the SER, of course) does this adequately protect the server from unauthorized users or is there something else to do? 2. according to the wiki the autocreatepeer creates peers based on the global variables. some variables, like dtmfmode, for example, are listed as belonging to individual peers. if i set dtmfmode, or qualify, or any of the others listed as individual variables, in [general] will the autocreatepeer use them? I suppose i could write a script to automatically generate peers for asterisk from SER's DB, (along the lines of the current retrieve_sip_conf_from_mysql.pl) but having duplicate SIP client entries seems kind of inelegant. And, of course, if i'm missing something basic conceptually, i'd be grateful if someone could point that out to me as well. any help is appreciated, thanks- yair ___ 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 'Under the Hood' Project
I'd thought I'd been through the whole Zapata Telephony Site. Could you e-mail back and point to the specific links you had in mind? Start with http://www.zapatatelephony.org/philos.html and dive into http://www.zapatatelephony.org/project.html and then into http://www.zapatatelephony.org/conf.html ___ 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] 'connecting' voip-numbers to our Asterisk
Hi! Sample configuration or other documentation from the provider? Hmm, haven't received any! :-/ all I got was username password... Is there a way (perhaps with sipsak?) to determine what kind of server/system they are running? If their system is not IAX-compatible, what are my options then for routing incoming, outgoing or both via this voip-provider? Greetings, Evert Benjamin on Asterisk Mailing Lists wrote: On Wed, 08 Sep 2004 10:08:00 +0200, Evert Meulie [EMAIL PROTECTED] wrote: I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf * [ip-incoming] exten = ,1,Dial(106,20,r) * /etc/asterisk/iax.conf * register = :[EMAIL PROTECTED] * This should be all I need to let incoming calls on ring on extension 106, right? No. First of all, let me ask you this... Are you sure that this provider supports IAX? I am asking because the Cisco 7960 doesn't do IAX, so you wouldn't have been using IAX when connecting directly. Second, if your provider does support IAX, then you will also need to set up a peer for incoming connections and send the calls to your incoming context, like so ... [iaxprovider] type=user username=888 secret=blah host=iax.provider.com qualify=yes disallow=all allow=whatever-codec-they-support context=incoming-from-iaxprovider this may or may not work depending on how your provider will try to connect to you. For example, FWD will always come in as user iaxfwd, so if you don't define your inbound peer as [iaxfwd] it won't work. Also, some providers use passwords, others use RSA keys. but assuming that the above matches the way in which your provider expects to connect to you, then you will still need an incoming context in extensions.conf named the same way as whatever comes after the context= setting. Even that may not be enough depending on how your proider presents the call to you. They may come in using your username or number, but they may as well use an account code or simply s. You will have to check out the sample configuration or whatever other documentation they provide. The chance is that somebody on this list is using the same provider, so you may tell us what provider you are using and somebody may then share their configuration with you. Also, the Wiki may have a sample configuration for the provider you are using. I always use the IAX debug command on the console to find out how an IAX peer comes in. Simply enter the command iax2 debug on the Asterisk console, then make a test call and see what the debug output says. It's pretty self explanatory. Use the command iax no debug to turn debugging off again. rgds benjk ___ 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] X-Lite Meetme problem
And the same problem with Grandstream HandyTone-286 as well On Wed, 2004-09-08 at 11:43, Vladyslav wrote: HI! Have a weird problem with X-lite Meetme. When X-Lite user are join to conference room NOT first one, than X-Lite user do not hear anything. This problem gone when X-Lite user get into conference room first (when nobody there). sip.conf [104] context=VoIP-only type=friend username=104 secret=test host=dynamic dtmfmode=rfc2833 mailbox=104 canreinvite=no disallow=all allow=ulaw ;allow=alaw ;allow=gsm On * console have such messages: when X-Lite using ULAW: Sep 8 09:47:17 WARNING[1233853360]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 64/4) on x-lite ALAW Sep 8 09:49:12 WARNING[1235344304]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 8 (read/write = 64/8) on x-lite GSM Sep 8 09:51:50 WARNING[1233804208]: chan_sip.c:1838 sip_write: Asked to transmit frame type 64, while native formats is 2 (read/write = 64/2) Please advice. -- Best regards Vlad ___ 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] chan_capi error
Hello! Since i was not able to compile chan_capi 3.5 on either Fedora2, Debian stable/testing/unstable i decided to use the normal sources, and then patch chan_capi with the debian patch. Now i can compile chan_capi woth no errors. When i start asterisk on Fedora2 (2.6.5-1.358smp), i get this: . . . == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [res_features.so] = (Call Parking Resource) == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [chan_capi.so]Sep 8 11:42:42 WARNING[-150155136]: loader.c:248 ast_load_resource: /usr/lib/asterisk/modules/chan_capi.so: undefined symbol: __use_ast_pthread_create_instead__Sep 8 11:42:42 WARNING[-150155136]: loader.c:380 load_modules: Loading module chan_capi.so failed! The /usr/lib/asterisk/modules/chan_capi.so module exists. Whats wrong here? What does the error mean? Thanks, Mario ___ 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] Maximum tollerable lag/jitter for IAX2 w/ojitterbuffer enabled?
On Tue, 7 Sep 2004, Chris Shaw wrote: All calls are running as GSM, even though g.729 is also an 'allowed' codec (w/5 licenses installed). During an average call 'iax2 show channels' provides: Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter JitBuf Format 10.0.40.140 astpbx-woo 2/2 5/6 00040ms 0036ms ms GSM If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP doesn't just disconnect sockets unless it recieves a RESET or a FINISHED or there's a timeout (usually like 5 minutes or more depending on your TCP/IP stack). Needless to say that to disconnect a TCP connection, that would have to be one hell of a lag spike... * must be actively disconnecting the connection I've heard the jitter buffer is a bit buggy, have you tried turning it off completely? IAX runs over UDP. Most probably this users' problem is something to do with NAT between the two endpoints that is killing the connection between the endpoints. The jitterbuffer is disabled - see the ms in the JitBuf column. Steve ___ 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] Maximum tollerable lag/jitter for IAX2 w/oji tterbuffer enabled?
On Tue, 7 Sep 2004, Kris Boutilier wrote: Reproducing it is part of the problem. I've been getting user reports on and off for some time but I can't find anything out of the ordinary - initially this was looking like a Voicemail bug as many people were getting cut off while leaving messages. How would I debug the precise drop condition? I've Googled for more information on 'iax2 debug' but come up naught. Run Asterisk with debugging turned on - see my various posts here explaining how to capture it all in /var/log/asterisk/debug. That will reveal all. But I suspect you got some NAT between the end points, and that NAT is messing up and breaking the communications? Steve ___ 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] H323 Control Protocol Error
Hi there ! I searched the whole web to find some helping information about H323 Control Protocol, but there is no way to find that information. We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2 + 'asterisk-oh323_1.5 channel driver + wrapper' and configured the dialplan for using our H323 Endpoints which are ip200 Innovaphones. Besides, we also use Gnomemeeting but don't care it's not the problem, I think ! The whole endpoints are registered on an ip400 Gatekeeper which routes every call to asterisk, and asterisk processes the Dialplan and sends the call back to the ip400 and to the correct Endpoint. With this configuration the Endpoints can dial each other above the Gatekeeper and Dial Plan. ;-) well pretty fine - the only damped thing is every call loses connection after 30 sec because of a a H323 control protocol error . this is the asterisk output while phoneing : ### *CLI -- Executing Dial(OH323/R1, OH323/[EMAIL PROTECTED]:1720|15) in new stack -- Called [EMAIL PROTECTED]:1720 -- OH323/L13468 answered OH323/R1 *** [ip$x.x.x.x:2507/1] H.323 CONTROL PROTOCOL ERROR (Capability Exchange) *** [ip$x.x.x.x:2507/1] H.323 CONTROL PROTOCOL ERROR (Master-Slave Determination) *CLI Sep 2 13:57:15 ERROR[294931]: chan_oh323.c:1212 oh323_hangup: OH323/L13468: Failed to hangup channel (timeout). -- Hungup 'OH323/L13468' == Spawn extension (buero, 3020, 1) exited non-zero on 'OH323/R1' -- Hungup 'OH323/R1' *CLI ### I hope you can help me and the whole asterisk community to solve this problem Hopefully, and waiting for response greets alex ___ 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] Maximum tollerable lag/jitter for IAX2 w/o j itterbuffer enabled?
On Tue, 7 Sep 2004, Kris Boutilier wrote: The only thing that springs to mind, if it's all UDP driven, is a lack of retry handler for the UDP handoff acknowledgement? I'm averaging about a 0.5% collision rate on this network (half-duplex 10Base-T)... My IAX connections soldier on over links with latency varying from 30msec to 1000msec+, with packet loss up to 10%. You hear the effects, but you don't lose your connection. Without the jitter buffer, whatever chan_iax2.c receives, you hear. If you don't hear anything, either the sender didn't send anything, or your network connection is gone. Steve ___ 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] Assigning a higher irq to a digium card
Hi, I have on my dual opteron (64 bit mode, linux) the problem, that sometimes read errors (unknown error 500) occur. This was already discussed on some asterisk list, and the solution seems to be to put the digium card on the highest interrupt level. Unfortunately I don't know howto. Applying an irq parameter when loading the module don't work: wct4xxp: Unknown parameter `irq' Within the bios menu I can't find any appropriate mean. But I assume, that there is a operating system mean, in order to assign the digium card another interrupt. Thanks for any hints! Roger. cat /proc/interrupts CPU0 CPU1 0: 92446503 0IO-APIC-edge timer 2: 0 0 XT-PIC cascade 8: 0 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 35414 0IO-APIC-edge ide0 15: 28 0IO-APIC-edge ide1 19: 0 0 IO-APIC-level ohci_hcd, ohci_hcd 28: 92169476 0 IO-APIC-level t4xxp 29: 924252 0 IO-APIC-level eth0 30:2516959 0 IO-APIC-level eth1 NMI: 8011699 LOC: 92432790 92433436 ERR: 0 MIS: 0 ___ 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 and the number of rings
On Wed, 2004-09-08 at 13:43, HengWee Chin wrote: Hi all, I have the following setup PSTN - ASTERISK - IVR (using dialogic card) 1) Caller id information is presented to asterisk during the first and second ring. 2) Hence, Asterisk waits for 2 rings before pickup the call and forwarding to the appropriate FXS port. 3) The IVR application also waits for 2 rings before picking up the call to get the caller id. 4) Hence any caller calling to the IVR will have to wait for 4 rings before he is serviced. This is too long. 5) Anyone have any idea how can I reduced the number of rings and still have caller id available to IVR? AFAIK, if asterisk has already waited 2 rings for the callerid, then why would the IVR need to wait as well? You shouldn't need to wait in the dialplan as well. Try removing that, and you should still have callerid. 6) If I were to switch PRI ISDN, would I still have the same problem? Yup, we use an E1, and can answer immediately (no rings at all) and still have callerid. This is ultimately the best solution, regardless of your application. The only time you don't use it, is if you can't afford it. AFAICT, the second best solution is to use chan_capi/zaphfc with supported BRI cards. Just my 0.02c worth Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] zaphfc strange errors
Hi i've an hfc-s card with last bristuff installed at cli i'm receiving: Sep 8 12:35:20 WARNING[1109552048]: chan_zap.c:6902 zt_pri_error: PRI: !! Got a UA, but i'm in state 1 received TEI check request for TEI = 77 what is causing them? 10x Maurizio ___ 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] sendmailhostname
Good day all I'm just wondering for interest sake I have a box,hostname=myname.co.za,running sendmail If I send mail to [EMAIL PROTECTED] it try to deliver to the box,witch does not have the mail box.How do I tell sendmail that it should send mail to myname.co.za's mailserver. I know how easy it is to change the name but there's a lot of reasons why we can.It is not in the local-hostnames file either. Thanks Altus ___ 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] sendmailhostname
get a book on DNS andlookup MX records or look on yolinux for a tutorial Altus Snyman wrote: Good day all I'm just wondering for interest sake I have a box,hostname=myname.co.za,running sendmail If I send mail to [EMAIL PROTECTED] it try to deliver to the box,witch does not have the mail box.How do I tell sendmail that it should send mail to myname.co.za's mailserver. I know how easy it is to change the name but there's a lot of reasons why we can.It is not in the local-hostnames file either. Thanks Altus ___ 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] 'connecting' voip-numbers to our Asterisk - update
Hi! It turns out my provider uses the Micronet SIP server. Any possibilies to let this one interface with Asterisk? Regards, Evert Evert Meulie wrote: Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf * [ip-incoming] exten = ,1,Dial(106,20,r) * /etc/asterisk/iax.conf * register = :[EMAIL PROTECTED] * This should be all I need to let incoming calls on ring on extension 106, right? Regards, Evert ___ 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] asterisk console from xinetd?
I'm trying to set up xinetd to run an asterisk console on a tcp port.
So far I've added a file in /etc/xinetd.d/ like:
service actl
{
disable = no
socket_type = stream
protocol= tcp
port= 1234
wait= no
user= root
server = /usr/sbin/asterisk
server_args = -r -n
log_on_failure += USERID
}
After adding actl to /etc/services and restarting xinetd it reports
one new service. When connecting to port 1234 on 127.0.0.1 (iptables
preventing remote hosts from accessing this service) I see the CLI prompt
repeating over and over with no line breaks.
Any idea how to prevent the looping please?
Thanks,
Mark.
p.s. Why am I doing this? We have an application that already knows
how to talk to other things via TCP sockets and we'd like to make it
talk to Asterisk too. The network between the two servers is trusted
so sending stuff clear-text isn't a problem.
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[Asterisk-Users] SIP and */#
hi all I'm trying to setup call divertion with the standard *21*numbertodivertto# etc but... When I dial such a number from a SIP client, it generally works quite badly most of the ones I've tried can handle *, but none, or at least few, can handle # Is this a SIP protocol weakness, or what is this? roy ___ 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 and */#
On Wed, 8 Sep 2004, Roy Sigurd Karlsbakk wrote: hi all I'm trying to setup call divertion with the standard *21*numbertodivertto# etc but... When I dial such a number from a SIP client, it generally works quite badly most of the ones I've tried can handle *, but none, or at least few, can handle # Is this a SIP protocol weakness, or what is this? I noticed that X-Lite sends the # like URL-encoded For example to dial $, X-Lite sends To: sip:[EMAIL PROTECTED];tag=as3355637e Asterisk obviously doesn't convert that back. I don't know whether it should, or whether X-Lite shouldn't encode like that. Probably, Asterisk should be fixed. Steve ___ 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] Maximum tollerable lag/jitter for IAX2w/ojitterbuffer enabled?
On Sep 7, 2004, at 7:15 PM, Chris wrote: Asterisk never ever uses TCP for IAX or IAX2. It's ALWAYS UDP. I don't believe Asterisk supports SIP over TCP either. Heck, the manager port is the only thing that uses TCP that I know of with Asterisk. Hmmm I wonder why I had the impression that it was TCP... You're right... Still, because UDP is connectionless and stateless it still won't disconnect on a LAG spike. I've had UDP sit there and send packets at my machine 2 hours after it had been shut down... You *really* don't want it to be TCP. Think about how TCP reacts to packet loss--it keeps retransmitting the dropped packet and delays everything after it until the dropped packet arrives. Now imagine what that'd do to phone calls--among other things, you'd have ever-increasing delay times, and there's no way to catch back up--after even a small amount of packet loss, the call would be unusable. TCP's a wonderful protocol, just not for real-time traffic. Scott ___ 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 without ZAP?
On Sep 7, 2004, at 4:43 PM, Chris Shaw wrote: - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Tuesday, September 07, 2004 4:39 PM Subject: RE: [Asterisk-Users] MeetMe without ZAP? Matthew Boehm wrote: Since I am using a SMP machine without USB ports does that mean I am fuX0red and can't run MeetMe at all? You can try the zaprtc (search for a link), or go out to Staples/OfficeDepot/BestBuy and pick up a PCI USB adapter. It must be a UHCI USB adapter though and that's not usually written on the box anywhere! :) Yeah, most of them are probably OHCI or EHCI. Scott ___ 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] asterisk+chan_h323+redhat9 troubles
hi,
i had asterisk and gnugk running on fedora core 2. it worked quite well. then, i needed
to change to red hat 9, and i'm experiencing troubles with h.323 :-( making a call from
a h.323 phone (innovaphone) does not work, and dial-in also doesn't. below
is an excerpt of what happens, when i try to dial-in my extension (126). it takes
about 10(!) seconds, until the 'Called 126' line appears.
the phone starts to ring a few seconds later. and when the caller drops, the phone
continues to ring, until asterisk crashes.
the gnugk runs on the same server as asterisk does.
has anyone an idea of what could be wrong here?
pm
-- Executing Macro(Zap/1-1, stdexten|126|H323) in new stack
-- Accepting call from '69911590527' to '126' on channel 0/1, span 1
-- Executing DBget(Zap/1-1, temp=CFIM/126) in new stack
-- DBget: varname=temp, family=CFIM, key=126
-- DBget: Value not found in database.
-- Executing GotoIf(Zap/1-1, 1 ? 200 : 300) in new stack
-- Goto (macro-stdexten,s,200)
-- Executing Dial(Zap/1-1, H323/126H323/426|15) in new stack
phoney*CLI Allowed Codecs:
Table:
G.711-ALaw-64k{sw} 1
Set:
0:
0:
G.711-ALaw-64k{sw} 1
-- Making call to 126 using gatekeeper.
1:33.909 ThreadID=0x00060013 h323ep.cxx(1323) H323
Making call to: 126
== New H.323 Connection created.
-- 69911590527 is calling host 126
-- Call token is ip$localhost/3103
-- Call reference is 3103
-- Called 126
phoney*CLI Allowed Codecs:
Table:
G.711-ALaw-64k{sw} 1
Set:
0:
0:
G.711-ALaw-64k{sw} 1
-- Making call to 426 using gatekeeper.
1:39.641 ThreadID=0x00060013 h323ep.cxx(1323) H323
Making call to: 426
2004-09-08 14:50:35 WARNING[163850]: chan_zap.c:6962 zt_pri_error: PRI: !!
Got reject for frame 1, but we have nothing -- resetting!
2004-09-08 14:50:35 WARNING[163850]: chan_zap.c:6962 zt_pri_error: PRI: !!
Got reject for frame 1, but we have nothing -- resetting!
phoney*CLI == New H.323 Connection created.
-- 69911590527 is calling host 426
-- Call token is ip$localhost/3104
-- Call reference is 3104
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[Asterisk-Users] accept DTMF while beeing in a queue
Hello, I would like to know, if it is possible to accept DTMF signals from a caller while he is in a queue. I would like to accomplish something like this: 1) The caller is in the queue. 2) The caller dials 123. 3) The caller is sent to extension 123. just for your information: When the caller is in the queue and sends a DTMF signal I see this message: DEBUG[327698]: chan_zap.c:3955 zt_read: DTMF digit: 5 on Zap/4-1 Regards, Fabian Müller ___ 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 and */#
filed as 0002399 On 8. sep. 2004, at 14.25, [EMAIL PROTECTED] wrote: On Wed, 8 Sep 2004, Roy Sigurd Karlsbakk wrote: hi all I'm trying to setup call divertion with the standard *21*numbertodivertto# etc but... When I dial such a number from a SIP client, it generally works quite badly most of the ones I've tried can handle *, but none, or at least few, can handle # Is this a SIP protocol weakness, or what is this? I noticed that X-Lite sends the # like URL-encoded For example to dial $, X-Lite sends To: sip:[EMAIL PROTECTED];tag=as3355637e Asterisk obviously doesn't convert that back. I don't know whether it should, or whether X-Lite shouldn't encode like that. Probably, Asterisk should be fixed. Steve ___ 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] asterisk console from xinetd?
Hello, On Wed, 8 Sep 2004 12:54:50 +0100 (BST), Mark Turner [EMAIL PROTECTED] wrote: I'm trying to set up xinetd to run an asterisk console on a tcp port. So far I've added a file in /etc/xinetd.d/ like: snip After adding actl to /etc/services and restarting xinetd it reports one new service. When connecting to port 1234 on 127.0.0.1 (iptables preventing remote hosts from accessing this service) I see the CLI prompt repeating over and over with no line breaks. Any idea how to prevent the looping please? p.s. Why am I doing this? We have an application that already knows how to talk to other things via TCP sockets and we'd like to make it talk to Asterisk too. The network between the two servers is trusted so sending stuff clear-text isn't a problem. Did you try asterisk manager? You can execute all of the cli commands and much more. Just enable it in /etc/asterisk/manager.conf and read manager.txt in the asterisk docs directory. -- Nicolás Gudiño Buenos Aires - Argentina ___ 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 without ZAP?
It specifically says here: http://www.voip-info.org/wiki-Asterisk+timer that zaprtc cannot be used with an SMP machine. What is this timer used for that it is so important? Matthew - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Tuesday, September 07, 2004 6:39 PM Subject: RE: [Asterisk-Users] MeetMe without ZAP? Matthew Boehm wrote: Since I am using a SMP machine without USB ports does that mean I am fuX0red and can't run MeetMe at all? You can try the zaprtc (search for a link), or go out to Staples/OfficeDepot/BestBuy and pick up a PCI USB adapter. - Andrew Thompson http://aktzero.com/ ___ 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] Problem playing file with G729A
Hi, I tried to play the standard demo-echotest file !. It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error: Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format: Unable to find a path from GSM to G729A Sep 8 14:58:33 WARNING[-182461520]: file.c:779 ast_streamfile: Unable to open demo-echotest (format G729A): No such file or directory Sep 8 14:58:33 WARNING[-182461520]: app_playback.c:83 playback_exec: ast_streamfile failed on SIP/media-gw-45.utanet.at-097eb490 for demo-echotest And i hear nothing ! - What have i done wrong ? What does it mean: Unable to find a path from GSM to G729A ?? (the file demo-echotest.gsm exists!!) br Johannes ___ 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] Maximum tollerable lag/jitter for IAX2 w/o j itterbuffer enabled?
On Tuesday 07 September 2004 20:55, Kris Boutilier wrote: The arrangement right now has: PSTN Trunks Stations - Nortel Norstar#1 -CT1- Asterisk#1 -IAX2- Asterisk#2 -CT1- Nortel Nortstar#2 - Stations The Asterisk boxes provide Voicemail to their sites Norstars and intersite calls over IAX. Local Voicemail works flawlessly at each site but there How are you getting MWI to light up on the digital phones? I was going to start screwing about with MCDN to try and achieve this (turning the MICS into nothing more than a digitla phone driver) but haven't started yet. -A. ___ 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] sendmailhostname
Hello, On Wed, 8 Sep 2004 13:10:48 +0200, Altus Snyman [EMAIL PROTECTED] wrote: Good day all I'm just wondering for interest sake I have a box,hostname=myname.co.za,running sendmail If I send mail to [EMAIL PROTECTED] it try to deliver to the box,witch does not have the mail box.How do I tell sendmail that it should send mail to myname.co.za 's mailserver. I know how easy it is to change the name but there's a lot of reasons why we can.It is not in the local-hostnames file either. Thanks Altus If your machine will never receive local mail, you can setup a 'SMART HOST' in sendmail, then, all mail will be relayed through that host. If you use redhat/fedora, you can add the smart host file in: /etc/mail/sendmail.mc just add the line: define(`SMART_HOST',`your.smtp.relay.server.or.ip.address')dnl Then you have to generate the /etc/sendmail/sendmail.cf with the command m4 /etc/mail/sendmail.mc /etc/mail/sendmail.cf Restart sendmail and you are done. Just backup sendmail.cf firts just in case something goes wrong. Best regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ 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] Problems with length of voicemail
Yes, I *am* using BROADVOICE, thanks for responses. Sure enough. If I dial in via my Washington number (ipkall), I don't have the problem. Interesting. Well, BV has a very good tech that seems to be very familiar with Asterisk. I'll see if he has any ideas how to deal with the issue. Sorry I didn't catch the earlier thread From: [EMAIL PROTECTED] on behalf of hank smith Sent: Wed 9/8/2004 03:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problems with length of voicemail I had that problem when I was running asterisk on my linux box before it went down so you aren't the only one having that problem - Original Message - From: Marty Mastera mailto:[EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion mailto:[EMAIL PROTECTED] Sent: Tuesday, September 07, 2004 10:50 PM Subject: RE: [Asterisk-Users] Problems with length of voicemail I wonder if anyone else's Asterisk box drops the connection to voicemail after 30 secs even when the maxmessage parameter is set to 180 (3 mins). Here is the general section of my voicemail: Roger, There has been very recent discussion regarding this topic exactly...specifically when using BroadVoice as a sip provider. Calls to your BroadVoice DID that end up in VM terminate after 30 seconds The current theory is that during VM recording, * doesn't send any audio packets back to BroadVoice...after 30 seconds BroadVoice thinks that the connection has been lost and terminates the call...(I'm paraphrasing the thread that recently appeared on this topic, forgive me if this isn't completely accurate) Assuming that this is correct, you could be using BroadVoice, or another provider who disconnects after not receiving audio for some period of time... Hope that helps, Marty ___ 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 My Inbox is protected by SPAMfighter 983 spam mails have been blocked so far. Download free SPAMfighter http://www.spamfighter.com/functions/split.aspx?gid=8 today! winmail.dat___ 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] Maximum tollerable lag/jitter for IAX2 w/ojitterbuffer enabled?
On Tuesday 07 September 2004 19:39, Chris Shaw wrote: If you can reproduce it, this smells like a bug... IAX runs over TCP and TCP doesn't just disconnect sockets unless it recieves a RESET or a FINISHED or there's a timeout (usually like 5 minutes or more depending on your TCP/IP stack). Needless to say that to disconnect a TCP connection, that would have to be one hell of a lag spike... * must be actively disconnecting the connection uh, no. IAX2 is UDP, not TCP. I don't know of any VOIP audio transports that are TCP based. -A. ___ 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 and */#
After small review of the chan_sip.c you should turn on pedantic sipchecking pedantic=yes in sip.conf [general] bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Wednesday, September 08, 2004 8:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP and */# filed as 0002399 On 8. sep. 2004, at 14.25, [EMAIL PROTECTED] wrote: On Wed, 8 Sep 2004, Roy Sigurd Karlsbakk wrote: hi all I'm trying to setup call divertion with the standard *21*numbertodivertto# etc but... When I dial such a number from a SIP client, it generally works quite badly most of the ones I've tried can handle *, but none, or at least few, can handle # Is this a SIP protocol weakness, or what is this? I noticed that X-Lite sends the # like URL-encoded For example to dial $, X-Lite sends To: sip:[EMAIL PROTECTED];tag=as3355637e Asterisk obviously doesn't convert that back. I don't know whether it should, or whether X-Lite shouldn't encode like that. Probably, Asterisk should be fixed. Steve ___ 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] astcc dont write to the table cdrs or cards
Hi, I did a cvs update on 03 Sep. How do I find out all available variables (to agi) in a particular code version. I tried show agi get variable, but that wouldnt give me much info. Cheers dh -Original Message- From: Areski [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 08, 2004 1:38 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] astcc dont write to the table cdrs or cards Variables DIALSTATUS: added to CVS head in june/july 2004! What is your CVS version? Areski On Wed, 2004-09-08 at 03:44, Doug Harris wrote: Hi, I have set-up astcc with outgoing sip channel. Call processing works fine but after the call tables, CDR and Cards does not get updated. At the beginning it goes to the database and fetch card details and correctly provides the card balance etc. Also it indeed write the inuse field (so writing and reading from database works fine). I've inserted a break point as such in the code; $dialstatus = $AGI-get_variable(DIALSTATUS); print STDERR dial status $dialstatus\n; It seems like dialstatus is not returned (which prints nothing). So obviously later part of the agi does not go through database updating portion (which only happens if dialstatus = Answerd). I am using deadagi to call the astcc.agi script as explained. Can someone explain why this happens ? Cheers dh __ ___ 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] Help needed!
Hi all, I am an MTech student and currently working on a project on GSM air interface. I am making use of Asterisk soft PBX. I am stuck at a point regarding this. As far as I understood from the available Asterisk documentation that Asterisk can easily plug into it the various programming interfaces and different codecs in it can seemlessly talk to one another. Asterisk has a codec translator API for GSM. Is it possible to make Asterisk directly communicate with a GSM air interface module thourgh the GSM codec API ? That means that i would be making a call from IP phone that wil be routed through the asterisk to the GSM interface. If it is possible, where should i make the necessay changes to enable this interworking. Kindly help. Any kind of suggestions are welcome. Renu Rangnekar ___ 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] Polycon IP 300 SIP vs Grandstream BT-101 Deployment
Hi, I have just completed the deployment of a couple of Grandstream phones (for internal IP use) and was wondering how much harder it would be to deploy a Polycom IP 300 phone. The Grandstream was quite easy to deploy and gives us good voice quality over DSL, however from some of the previous posts I am see that some people had troubles with the Polycom 300. The variant I am looking at purchasing is a SIP variant. Any pointers / comments would be greatly appreciated. Kind Regards Stuart ___ 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] Where to post SuSE 9.x startup script?
Hi all, I just modified one of the startup scripts provided on the tarball to fit on my SuSE 9.x system to start/stop Asterisk when the system boots or goes down. Maybe I'm overseeing the answer but could't find where to post/(cvs)upload the changes I made... TIA, Martin ___ 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 needed!
On Wed, 8 Sep 2004 18:58:11 +0530, Renu Rangnekar [EMAIL PROTECTED] wrote: As far as I understood from the available Asterisk documentation that Asterisk can easily plug into it the various programming interfaces and different codecs in it can seemlessly talk to one another. Asterisk has a codec translator API for GSM. Is it possible to make Asterisk directly communicate with a GSM air interface module thourgh the GSM codec API ? That means that i would be making a call from IP phone that wil be routed through the asterisk to the GSM interface. If it is possible, where should i make the necessay changes to enable this interworking. It will take a lot more than the codec translator and changes to make Asterisk talk to a GSM BTS. You would have to implement a significant part of the GSM MAP protocol and an SS7 stack. rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ 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] PRI issue
Hi, I recompiled asterisk today from CVS and Ive been having a number of problems, Ive read the deadlock page on the wiki and some of it sounds like that, however, the latest issue were having it that sometimes Asterisk doesnt seem to know the PRI channel has dropped, and assumes its still busy. However, that same channel can be used to make an outgoing call?! Has anyone experienced anything similar? Regards, Ben ___ 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] accept DTMF while beeing in a queue
Hello Fabian, Wednesday, September 8, 2004, 5:14:10 PM, you wrote: FM I would like to know, if it is possible to accept DTMF signals from a FM caller while he is in a queue. A context may be specified, in which if the user types a SINGLE digit extension while they are in the queue, they will be taken out of the queue and sent to that extension in this context. See queues.conf. -- Best regards, Olegmailto:[EMAIL PROTECTED] ___ 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 playing file with G729A
Hello Johannes, Wednesday, September 8, 2004, 5:21:53 PM, you wrote: JH Unable to find a path from GSM to G729A Use Google. You'll need a license for G.729. http://www.digium.com/index.php?menu=asterisk_g729 -- Best regards, Olegmailto:[EMAIL PROTECTED] ___ 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] WellGate 3504A with Asterisk SIP authentication and config
hey * folk, am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind SIP channel to context = default ; Default context for incoming calls [1235] host = dynamic secret = somepass context = default type = friend amaflags = billing accountcode = SIPUSER disallow=all allow=ulaw allow=alaw [1234] host = dynamic secret = somepass context = default type = friend amaflags = billing accountcode = SIPUSER disallow=all allow=ulaw allow=alaw and on the wellgate i've set the following under SIP information: Run Mode: Proxy Primary Proxy IP Address: 192.168.0.200 this is the IP addy of the asterisk server Secondary Proxy IP Address: Outbound Proxy: Proxy port: 5060 Line 1 Number: 1234 Line 1 Account: 1234 Line 1 Password: somepass Line 2 Number: 1235 Line 2 Account: 1235 Line 2 Password: somepass Line 3 Number: 1236 Line 3 Account: Line 3 Password: Line 4 Number: 1237 Line 4 Account: Line 4 Password: SIP port: 5060 RTP Port: 16384 Expire: 60 however, only Line 1 from the wellgate succesfully authenticates with asterisk. picking up the analog handset attached to line 1 (1234) works fine with calls in and out working the way it should without any problems. Line 2 (and subsequently Lines 3 4) do not work at all. i've captured the output of 'sip debug' and have attached it below. looking at the output, it does seem that asterisk is rejecting the authentication for line 2 but it doesn't mention why. the obvious reasons (password mismatch et al) have been ruled out, of course. any help would be much appreciated. i've read mailing list archive of others being able to use the WellGate 3502 (which is just a 2xFXS port version) with asterisk. however no mention of VoIP protocols was mentioned as the wellgates traditionally supported H.323 but with firmware upgrade has been able to support SIP. (sip debug output begins) Sip read: REGISTER sip:192.168.0.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0 Max-Forwards: 70 Supported: replaces User-Agent: FXS_GW (4asipfxs.107a) Contact: sip:[EMAIL PROTECTED]:5060;expires=60 From: sip:[EMAIL PROTECTED] ;tag=c0a800ca-13c4-0-35c-48a3 To: sip:[EMAIL PROTECTED] Call-ID: c0a800ca-13c4-0-334-1c34 CSeq: 1 REGISTER Content-Length:0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.202 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0 From: sip:[EMAIL PROTECTED] ;tag=c0a800ca-13c4-0-35c-48a3 To: sip:[EMAIL PROTECTED];tag=as2b502f28 Call-ID: c0a800ca-13c4-0-334-1c34 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.202:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0 From: sip:[EMAIL PROTECTED] ;tag=c0a800ca-13c4-0-35c-48a3 To: sip:[EMAIL PROTECTED];tag=as2b502f28 Call-ID: c0a800ca-13c4-0-334-1c34 CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=00727441 Content-Length: 0 to 192.168.0.202:5060 Sip read: REGISTER sip:192.168.0.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc Max-Forwards: 70 Supported: replaces User-Agent: FXS_GW (4asipfxs.107a) Contact: sip:[EMAIL PROTECTED]:5060;expires=60 From: sip:[EMAIL PROTECTED] ;tag=c0a800ca-13c4-413ef95b-488-4bff To: sip:[EMAIL PROTECTED] Call-ID: c0a800ca-13c4-0-334-1c34 CSeq: 2 REGISTER Content-Length:0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.202 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc From: sip:[EMAIL PROTECTED] ;tag=c0a800ca-13c4-413ef95b-488-4bff To: sip:[EMAIL PROTECTED];tag=as2b502f28 Call-ID: c0a800ca-13c4-0-334-1c34 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.202:5060 Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-48d-3dcc From: sip:[EMAIL PROTECTED] ;tag=c0a800ca-13c4-413ef95b-488-4bff To: sip:[EMAIL PROTECTED];tag=as2b502f28 Call-ID: c0a800ca-13c4-0-334-1c34 CSeq: 2 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=7b6d7de6 Content-Length: 0 to 192.168.0.202:5060 Sip read: REGISTER sip:192.168.0.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-413ef95b-5b9-6ab8 Max-Forwards: 70
Re: [Asterisk-Users] Zaptel 'Under the Hood' Project
Holger Schurig wrote: I'd thought I'd been through the whole Zapata Telephony Site. Could you e-mail back and point to the specific links you had in mind? Start with http://www.zapatatelephony.org/philos.html and dive into http://www.zapatatelephony.org/project.html and then into http://www.zapatatelephony.org/conf.html ___ 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 The first two pages I've seen but I must admit the last is new to me. It must have been hiding in plain sight. Thanks! ___ 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] Where to post SuSE 9.x startup script?
I would be interested in the script. Did you do zaptel drivers too? On Wed, 2004-09-08 at 10:41, Martin Mielke wrote: Hi all, I just modified one of the startup scripts provided on the tarball to fit on my SuSE 9.x system to start/stop Asterisk when the system boots or goes down. Maybe I'm overseeing the answer but could't find where to post/(cvs)upload the changes I made... TIA, Martin ___ 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] OH323 Ignoring PROGRESS indication
Good time of day all! 1) I am trying to use as5300 and asterisk. As5300 sends calls to me. I get the following in * console: -- IAX2/magrathea/6 is making progress passing it to OH323/R27464 Sep 8 10:57:59 NOTICE[1140046640]: chan_oh323.c:1159 oh323_indicate: Ignoring PROGRESS indication. As5300 user does not hear anything, just silense instead of dial tones. My config is oh323.conf default config. 2) * logs CDR records with NO Answer and duration 0, but h323 shows duration 0. Why so? ___ 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 and the number of rings
Just a comment here. I built an * pbx on a Celeron 1.4ghz machine. Got all the dialplan and such working , then built a new server with an AMD 2.4g processor with a 500mhz front side buss. With the same Digium TDM cards and all analog incoming and outgoing. The celeron was not ringing out until the third incoming ring. The new server starts ringing inside just before the second ring hits the incoming analog port. Same version of * and same version of Suse Linux, better processer, better buss speed, and now a serial ata hard drive. So the speed of the server does also have some effect. Lyle - Original Message - From: Adam Goryachev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Wednesday, September 08, 2004 5:30 AM Subject: Re: [Asterisk-Users] Caller id and the number of rings On Wed, 2004-09-08 at 13:43, HengWee Chin wrote: Hi all, I have the following setup PSTN - ASTERISK - IVR (using dialogic card) 1) Caller id information is presented to asterisk during the first and second ring. 2) Hence, Asterisk waits for 2 rings before pickup the call and forwarding to the appropriate FXS port. 3) The IVR application also waits for 2 rings before picking up the call to get the caller id. 4) Hence any caller calling to the IVR will have to wait for 4 rings before he is serviced. This is too long. 5) Anyone have any idea how can I reduced the number of rings and still have caller id available to IVR? AFAIK, if asterisk has already waited 2 rings for the callerid, then why would the IVR need to wait as well? You shouldn't need to wait in the dialplan as well. Try removing that, and you should still have callerid. 6) If I were to switch PRI ISDN, would I still have the same problem? Yup, we use an E1, and can answer immediately (no rings at all) and still have callerid. This is ultimately the best solution, regardless of your application. The only time you don't use it, is if you can't afford it. AFAICT, the second best solution is to use chan_capi/zaphfc with supported BRI cards. Just my 0.02c worth Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] Where to post SuSE 9.x startup script?
Tony Nichols wrote: I would be interested in the script. OK. I'll send it off the list... Did you do zaptel drivers too? Nope ;) Martin ___ 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] Problems with length of voicemail
-Original Message--- Yes, I *am* using BROADVOICE, thanks for responses. Sure enough. If I dial in via my Washington number (ipkall), I don't have the problem. Interesting. Well, BV has a very good tech that seems to be very familiar with Asterisk. I'll see if he has any ideas how to deal with the issue. Sorry I didn't catch the earlier thread -Begin Reply-- Myself and several others have had this problem. (Anyone using an ITSP who uses BroadWorks I imagine). My current theory is that BroadWorks requires some acknowledgement that the sending side, Asterisk, is still there. Right now because * has no CNG (Comfort Noise Generation) or DTX (Discontinuous TX) support, it does not send anything back to the receiver until recording is finished... BroadWorks takes this as a sign that * has lost the connection and tears down the connection... (You can see this in *, it says User Hung Up). I did mean to say BroadWorks, that's the brand of VoIP switch that BroadVoice uses. I have a strong suspicion as stated earlier that any provider that uses BroadWorks WILL have this same problem... -Chris ___ 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] PRI issue
Ben, I ran into a similar issue on the 8/31 cvs, except it was backwards. Outbound calls would report a busy on the channel selected, yet a few minutes later the channel would be used for an inbound call. I had to revert back to my previous checkout from 8/16 to resolve the issue. The problems didn't break the channels completely, it happened probably every 5-10 minutes. Brian D'Arcy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills Sent: Wednesday, September 08, 2004 7:51 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI issue Hi, I recompiled asterisk today from CVS and I've been having a number of problems, I've read the deadlock page on the wiki and some of it sounds like that, however, the latest issue we're having it that sometimes Asterisk doesn't seem to know the PRI channel has dropped, and assumes it's still busy. However, that same channel can be used to make an outgoing call?! Has anyone experienced anything similar? Regards, Ben ___ 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] OH323 Ignoring PROGRESS indication
Try, in the 53 (depends on the SW version u're using voice call send-alert Also if you're using PRI trunks you can use, in the Serial interface,. isdn send-alerting -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Maxim Litnitsky Sent: Wednesday, September 08, 2004 12:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] OH323 Ignoring PROGRESS indication Good time of day all! 1) I am trying to use as5300 and asterisk. As5300 sends calls to me. I get the following in * console: -- IAX2/magrathea/6 is making progress passing it to OH323/R27464 Sep 8 10:57:59 NOTICE[1140046640]: chan_oh323.c:1159 oh323_indicate: Ignoring PROGRESS indication. As5300 user does not hear anything, just silense instead of dial tones. My config is oh323.conf default config. 2) * logs CDR records with NO Answer and duration 0, but h323 shows duration 0. Why so? ___ 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] Newbie: Only allow authenticated users to call
Update to latest CVS and/or put context=INVALID in [general] in sip.conf and in each peer/user/friend entry put in a correct context= line. On Wed, 2004-09-08 at 03:31, Henry Jensen wrote: I made the observation that I'm able to make a call with my SIP client (kphone) even when I'm not registered/authenticated. Of course, when I'm not registered at asterisk, people can't call me, but it's still a huge security hole, that unregistered Clients can make calls. Is there a way to tell asterisk to only allow registered clients making calls? I know about the Anti Ex Girlfriend function, but this is not what I want. Regrads, Henry ___ 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 -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ 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] PRI issue
I think this issues stems from the (in my case) wct4xxp driver. When updating libpri, I also updated zaptel, however, I'm unsure if I installed it correctly (i.e. updated to the newly compiled version). After stopping asterisk, doing rmmod wct4xxp, make install on zaptel and then restarting asterisk, so far, it seems to be working. I'm not 100% sure this is the problem, but it would seem this resolved the issue... time will tell :) Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian D'Arcy Sent: 08 September 2004 16:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] PRI issue Ben, I ran into a similar issue on the 8/31 cvs, except it was backwards. Outbound calls would report a busy on the channel selected, yet a few minutes later the channel would be used for an inbound call. I had to revert back to my previous checkout from 8/16 to resolve the issue. The problems didn't break the channels completely, it happened probably every 5-10 minutes. Brian D'Arcy From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Merrills Sent: Wednesday, September 08, 2004 7:51 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] PRI issue Hi, I recompiled asterisk today from CVS and I've been having a number of problems, I've read the deadlock page on the wiki and some of it sounds like that, however, the latest issue we're having it that sometimes Asterisk doesn't seem to know the PRI channel has dropped, and assumes it's still busy. However, that same channel can be used to make an outgoing call?! Has anyone experienced anything similar? Regards, Ben ___ 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] Where to post SuSE 9.x startup script?
Huddleston, Robert wrote: I'd like a startup script for redhat... should be just some small changes.. do you have one? It's already there... :-) Take a look at .../asterisk_v1_0_stable/contrib/init.d to find a file called rc.redhat.asterisk. This one should do the trick... ;) HTH, Martin ___ 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] WellGate 3504A with Asterisk SIP authentication and config
On 08/09/2004 20:29 Dinesh Nair said the following: am trying to configure a WellGate 3504A FXS SIP ATA (http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set up two SIP clients in sip.conf as follows: i forgot to mention that i'm running asterisk 0.90.0 on freebsd 4.10 and the exact same settings (unchanged) in sip.conf work with the Xlite and SJPhone SIP softphones. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ 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] Got *80 working ... now some Blacklist questions
Il 05:41, mercoledì 08 settembre 2004, Steve Maroney ha scritto: On my default asterisk installation, *80 didn't work until I modified the source to move call pickup to *9. I wasn't sure what I was doing but *80 works now. Except I thought *80 would play some voice prompts that gave the option to add the last caller to the black list as well as other options. Instead I just get a studer dial tone after the last caller gets added to the database. I've opened a bug one month ago http://bugs.digium.com/bug_view_page.php?bug_id=0002247 that involve your problem. Diego ___ 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] Directory command assistance
I have searched through the wiki, but I was unable to find the information that I desire. I am trying to implement the Directory command within my Asterisk configuration. What information is passed back when a name is successfully found? Since Asterisk is being used as an automated attendant and voicemail solution for our PBX, configuration would be a little different from users who have VoIP phones. For all calls that come in, I need to use the following configuration: Flash() Wait(2) SendDTMF() - The extension of the user would be within the () Hangup How would I configure the dialplan to support the directory command? Thanks in advance for your assistance. ___ 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] Oh323, Please Help Newbie ;(
Hi, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=1 tcpEnd=2 udpStart=8000 udpEnd=8005 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=yes silenceSuppression=no jitterMin=20 jitterMax=500 ipTos=reliability outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 amaFlags=default accountCode=H323 [223];OpenPhone type=friend defaultip=193.25.30.223 username=223 context=default [register] . [codecs] codec=G711A frames=20 codec=G711U frames=20 codec=GSM0610 frames=4 /// /// Extension.conf [general] static=yes writeprotect=no [default] ;Xlite exten = 224,1,SetLanguage(de) exten = 224,2,Dial(SIP/xlite1,10) exten = 224,3,Voicemail(u224) exten = 224,102,Voicemail(b224) exten = 224,103,Hangup ;Openphone (H.323) exten = 223,1,SetLanguage(de) exten = 223,2,Dial(OH323/193.25.30.223,15) exten = 223,3,Voicemail(u223) exten = 223,102,Voicemail(b223) exten = 223,103,Hangup Thanks. ___ Gesendet von Yahoo! Mail - Jetzt mit 100MB Speicher kostenlos - Hier anmelden: http://mail.yahoo.de ___ 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] stale voicemail messages / greeting
I'm using Asterisk to read voicemail users out of a SQL database. I am
assigning users real phone numbers as their voicemail box. The problem is
that if I re-assign a phone number (say, 972-245-0001), the new user is
stuck with the old user's greeting and saved messages. What is the best way
to resolve this?
I don't want to use unique mailbox ids because my dialplan looks like this
in the incoming DID context
[incomingdids]
exten = _972245,1,setvar(boxnum=${EXTEN})
exten = _972245,2,VoiceMail(u${EXTEN})
exten = _972245,3,Hangup
exten = a,1,VoiceMailMain(${boxnum})
exten = a,2,Hangup
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[Asterisk-Users] Sending SIP call to Cisco 3660
All, i am new to asterisk, and I have been searching through the list and docs for examples on howto accomplish this, but i havent had much luck. 1. have asterisk answer when an unregistered cisco gateway send its a SIP call -- DONE. (using the demo samples i successfully get into the demo) 2. create an extension that i can dial (example: 1222) that will dial and outbound SIP call to another cisco gateway I have in melbourne australia (hooked to PSTN) 3. force the dialed number to be 613 (not 1222, these have no number expansion relation, just want to simply have a short extension to dial to get to a static number in australia). my cisco should take that and terminate it to the PSTN. I assume i have to make configs in sip.conf and exten.conf. I tried these entries. exten.conf exten = 1222,1,Dial(SIP/peter,r) sip.conf [peter] type=friend host=210.x.x.x username=613 but i get invalid extension message as soon as i dial '12' David Winter Senior Network Engineer Planet-Telecom, Inc. Tampa FL (813)901-5182 Office (813)864-3162 Direct (813)817-4204 Mobile (813)881-9762 Fax -- AIM: mobofool ICQ: 3563403 MSN:[EMAIL PROTECTED] Y!:vt_fool Joshua M. Thompson wrote: Oliver Breidenbach wrote: Hi there, what do I need to take into consideration if I want Asterisk to talk to a MySQL database on a different host to store CDR records? The cdr_addon_mysql module does not want to load and Asterisk claims that it "cannot open shared object". It compiled fine, however. It sounds like you don't have the MySQL client libraries properly installed, although you seem to have the headers since the module compiled. If your mysql client libs are not in /lib, /usr/lib or /usr/local/lib then you'll probably have to add whatever directory they are in to /etc/ld.so.conf and rerun ldconfig. ___ 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] zap: reroute incoming calls to dedicated channel
Hi ! I have a E100P and I would like to receive incoming calls on dedicated channels only. Is it possible to answer an incoming call request on channel 2-30 from the Telco with something like 'busy, use channel 1 instead' ? If this is possible, how could it be implemented / configured ? Br / Jan Terje ___ 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] Question on echo's for Canadian Asterisk users ...
Thanks for the tips ... Like you said, dealing with carrier is not going to get me anywhere. The only thing GT recommended was grounding the server chasis :P I turned the echo cancellation with the same parameters you used and It doesnt even make a difference. I dug further into the zaptel/Makefile To find the the echo cancellation algoriths: #KFLAGS+=-DECHO_CAN_STEVE KFLAGS+=-DECHO_CAN_STEVE2 #KFLAGS+=-DECHO_CAN_MARK None of these were listed at all in the Makefile, so I added them And tried a recompile. Still a bad echo. It is like the echo cancellation Is not even working. Is there a way to verify its active or not? Cheers, Paul Seniuk -Original Message- From: akohlsmith-asterisk [mailto:[EMAIL PROTECTED] Sent: September 4, 2004 3:51 PM To: asterisk-users Subject: Re: [Asterisk-Users] Question on echo's for Canadian Asterisk users ... On Saturday 04 September 2004 16:57, [EMAIL PROTECTED] wrote: Has anyone has issues with echo using a Wildcard with a PRI from a major Canadian Telco? (Bell, Telus, AllStream, Sprint, Group Telecom). I have a PRI with Bell Canada in Listowel, ON (519-291-). I have echo on some calls but not all -- it doesn't seem to have anything to do with what switch it's terminating on. Calling anywhere in Fordwich echos rather badly as do some Toronto numbers. The echo occurs on incoming and outgoing calls. (we only call out on the PRI for local, 800 and fax numbers). We are using a T1 from GT that is giving use annoying echos whenever a SIP/IAX2 client calls a local analog line. Calling cells phones is no issue since its digital. Regardless, there should be no issue with echo on a PRI at all. All that PRI gives you is one less hybrid in the circuit. That's it. NOC at GT is telling us that there is no echo cancellation enabled on this PRI. 'Talk to your rep' was the response I got To me thats crap, because they shouldnt be selling PRI's without this essential feature. Depending on who you talk to you will hear responses like 1. I have no idea what you're talking about. 2. We don't have echo cancellation hardware available on any PRI. 3. You must specifically provision the PRI with echo cancellation. I've found acceptable echo cancellation on the PRI with Asterisk's echo cancellation software on the TE405P with the following: - agressive cancellation echocancel=yes echocancelwhenbridged=yes echotraining=500 No need to worry about the echo canceller killing fax/data connections since just like the real echo cancellation hardware, asterisk will disable the echo cancel routines when it hears the correct disable tone on the line. You'll see something like zaptel Disabled echo canceller because of tone (tx) on channel 13. We were really having a lot of echo troubles but 20040831 CVS HEAD seems to have really helped, although it was certainly acceptable with 20040806 CVS HEAD. I haven't been able to locate a good hardware echo canceller on ebay yet (I keep missing the auctions). :-) -A. ___ 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] Intertex IX66
Hi, I get an Intertex IX66 and Im trying to connect my * behind this SIP router. I can register my Polycom phones on * but the sound on the phones is just one way. Someone fight with the same problem with this router? Any suggestions are really appreciated. Best regards, Chris HARIGA smime.p7s Description: S/MIME cryptographic signature ___ 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] accept DTMF while beeing in a queue
Oleg A. Arkhangelsky [EMAIL PROTECTED] writes: A context may be specified, in which if the user types a SINGLE digit extension while they are in the queue, they will be taken out of the queue and sent to that extension in this context. See queues.conf. Wow, thanks a lot Oleg. I overlooked that :-( Regards, Fabian Müller ___ 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] Asterisk with Primus Talkbroadband
Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense. I'm completely new to this. ___ 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] Question on echo's for Canadian Asterisk users ...
On Wednesday 08 September 2004 13:39, [EMAIL PROTECTED] wrote: None of these were listed at all in the Makefile, so I added them And tried a recompile. Still a bad echo. It is like the echo cancellation Is not even working. Is there a way to verify its active or not? It's not in the makefile, it's in zconfig.h. I've attached mine (which seems to work just fine). Hoping this doesn't break too many list rules, there are others who might benefit from a zconfig.h that seems to work very well. -A. /* * Zaptel configuration options * */ #ifndef _ZCONFIG_H #define _ZCONFIG_H #ifdef __KERNEL__ #include linux/config.h #include linux/version.h #endif /* Zaptel compile time options */ /* * Uncomment to disable calibration and/or DC/DC converter tests * (not generally recommended) */ /* #define NO_CALIBRATION */ /* #define NO_DCDC */ /* * Boost ring voltage (Higher ring voltage, takes more power) */ /* #define BOOST_RINGER */ /* * Define CONFIG_CALC_XLAW if you have a small number of channels and/or * a small level 2 cache, to optimize for few channels * */ /* #define CONFIG_CALC_XLAW */ /* * Define if you want MMX optimizations in zaptel * * Note: CONFIG_ZAPTEL_MMX is generally incompatible with AMD * processors and can cause system instability! * */ #define CONFIG_ZAPTEL_MMX /* * Pick your echo canceller: MARK2, MARK3, STEVE, or STEVE2 :) */ /* #define ECHO_CAN_STEVE */ /* #define ECHO_CAN_STEVE2 */ /* #define ECHO_CAN_MARK */ #define ECHO_CAN_MARK2 /* #define ECHO_CAN_MARK3 */ /* * Uncomment for aggressive residual echo supression under * MARK2 echo canceller */ /* #define AGGRESSIVE_SUPPRESSOR */ /* * Define to turn off the echo canceler disable tone detector, * which will cause zaptel to ignore the 2100 Hz echo cancel disable * tone. */ /* #define NO_ECHOCAN_DISABLE */ /* udev support */ #if LINUX_VERSION_CODE = KERNEL_VERSION(2,6,0) #define CONFIG_ZAP_UDEV #endif /* We now use the linux kernel config to detect which options to use */ /* You can still override them below */ #if defined(CONFIG_HDLC) || defined(CONFIG_HDLC_MODULE) /* #define CONFIG_ZAPATA_NET */ /* NEVER implicitly turn on ZAPATA_NET */ #if LINUX_VERSION_CODE = KERNEL_VERSION(2,4,20) #define CONFIG_OLD_HDLC_API #else #if LINUX_VERSION_CODE = KERNEL_VERSION(2,4,23) /* Starting with 2.4.23 the kernel hdlc api changed again */ /* Now we have to use hdlc_type_trans(skb, dev) instead of htons(ETH_P_HDLC) */ #define ZAP_HDLC_TYPE_TRANS #endif #if LINUX_VERSION_CODE = KERNEL_VERSION(2,6,3) #define HDLC_MAINTAINERS_ARE_MORE_STUPID_THAN_I_THOUGHT #endif #endif #endif #ifdef CONFIG_PPP #define CONFIG_ZAPATA_PPP #endif /* * Uncomment CONFIG_ZAPATA_NET to enable SyncPPP, CiscoHDLC, and Frame Relay * support. */ /* #define CONFIG_ZAPATA_NET */ /* * Uncomment CONFIG_OLD_HDLC_API if your are compiling with ZAPATA_NET * defined and you are using the old kernel HDLC interface (or if you get * an error about ETH_P_HDLC while compiling). */ /* #define CONFIG_OLD_HDLC_API */ /* * Uncomment for Generic PPP support (i.e. ZapRAS) */ /* #define CONFIG_ZAPATA_PPP */ /* * Uncomment to enable watchdog to monitor if interfaces * stop taking interrupts or otherwise misbehave */ /* #define CONFIG_ZAPTEL_WATCHDOG */ /* Tone zone info */ #define DEFAULT_TONE_ZONE 0 /* * Uncomment for Non-standard FXS groundstart start state (A=Low, B=Low) * particularly for CAC channel bank groundstart FXO ports. */ /* #define CONFIG_CAC_GROUNDSTART */ #endif ___ 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 Primus Talkbroadband
I have asked before and got no answers - I am still not clear as to why there is not an MGCP client as part of asterisk - is it a technical reason, no one else wants it, other ? It is my understanding primus is using mgcp, and therefore is unable to directly interface with asterisk, password or not. I would like to hear your experiences with quality though since I am considering switching to it anyway and run the dlink boxes into my channel bank for now, and figure out the software issues later. Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense. I'm completely new to this. ___ 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ 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] Polycom SIP 1.3.1 Reject Button
Hello, I recently upgraded to Sip 1.3.1 and noticed that the Reject Button is no longer appearent on the screen when a second incoming call comes in unless I press the hold button on the first call. Does anyone have a work around for this to reject a call while continuing to talk to the first party? I should also point out that I don't want it to be on *, as the situation varies from call to call. E.g. setting a count limit on a phone is not acceptable, as if the secretary is talking to someone from home, she can put them on hold and take the second call. Thanks, Brent D. Franks ___ 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] Driving MWI on Norstars (was Maximum tollera ble lag/jitter...)
At the moment we're not - the email notification from Comedian Mail has
been mostly sufficient. I do however have some Dialogic D/42-NS PBX
emulation cards and the plan is to use them to set and unset the MWI lamps
based on events pushed out of Asterisk.
They may be obsolete hardware but they came in real handy for extracting the
voicemail from the old StarTalk NAM too.
Take a look at the PBX Integration section of
http://resource.intel.com/telecom/support/releases/winnt/SR511FP1/onldoc/htm
lfiles/
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: September 8, 2004 6:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter
for IAX2 w/o
j itterbuffer enabled?
{clip}
How are you getting MWI to light up on the digital phones? I
was going to
start screwing about with MCDN to try and achieve this
(turning the MICS into
nothing more than a digitla phone driver) but haven't started yet.
-A.
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Re: [Asterisk-Users] Got *80 working ... now some Blacklist questions
Well I wouldn't look at my question as a problem. I thought I would get more functionality out of *80 after seeing other sound files in my sounds directory. Those voice prompts look useful so maybe I am missing something here. Thank you, Steve Maroney On Wed, 8 Sep 2004, Diego Ercolani wrote: Il 05:41, mercoledì 08 settembre 2004, Steve Maroney ha scritto: On my default asterisk installation, *80 didn't work until I modified the source to move call pickup to *9. I wasn't sure what I was doing but *80 works now. Except I thought *80 would play some voice prompts that gave the option to add the last caller to the black list as well as other options. Instead I just get a studer dial tone after the last caller gets added to the database. I've opened a bug one month ago http://bugs.digium.com/bug_view_page.php?bug_id=0002247 that involve your problem. Diego ___ 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] Question on echo's for Canadian Asterisk users ...
Yeah it looks to be the same setup as mine I am going to try out Mark3 and the Aggressive Suppresor as well. Paul Seniuk -Original Message- From: akohlsmith-asterisk [mailto:[EMAIL PROTECTED] Sent: September 8, 2004 12:34 PM To: asterisk-users Subject: Re: [Asterisk-Users] Question on echo's for Canadian Asterisk users ... On Wednesday 08 September 2004 13:39, [EMAIL PROTECTED] wrote: None of these were listed at all in the Makefile, so I added them And tried a recompile. Still a bad echo. It is like the echo cancellation Is not even working. Is there a way to verify its active or not? It's not in the makefile, it's in zconfig.h. I've attached mine (which seems to work just fine). Hoping this doesn't break too many list rules, there are others who might benefit from a zconfig.h that seems to work very well. -A. ___ 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] Maximum tollerable lag/jitter for IAX2 w/oji tterbuffer enabled?
Collecting debug today - got 8mb so far. Just turned up IAX2 debug so
expecting it to balloon.
There is no NAT involved, nor stateful firewalling etc. - this is a flat
10.0.0.0/24 subnet with one 10base-t hub and two 10base-t cable modems
(operating peer to peer) in between the endpoints. Bitrate is proven out at
not less than 512kbps using Iperf (http://dast.nlanr.net/Projects/Iperf/).
Jitter buffer, trunking and now native bridging (notransfer) code has been
intentionally disabled to reduce possible culprits. Hosts are running a load
average of between 0.00 and 0.02.
The call drop issue has been very sporadic and effectively unreproducable
from a testing perspective. At the moment I'm leaning in the direction of an
IAX2 ACK packet being dropped off the network - I've noticed about 0.5%
collisions on the wire (this being a half-duplex network) which seems to be
contributing to audible pops and clicks with the jitter buffer disabled.
I've also been getting occasional reports of far end echo on long distance
calls, but I suspect the ongoing thread about 'Question on echo's for
Canadian Asterisk users...' will get to the bottom of that one.
Does anyone know how robust/agressive the UDP retry code is in Asterisk?
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: September 8, 2004 3:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: 'Chris Shaw'
Subject: RE: [Asterisk-Users] Maximum tollerable lag/jitter for IAX2
w/oji tterbuffer enabled?
On Tue, 7 Sep 2004, Kris Boutilier wrote:
{clip}
Run Asterisk with debugging turned on - see my various posts here
explaining how to capture it all in /var/log/asterisk/debug.
That will reveal all.
But I suspect you got some NAT between the end points, and
that NAT is messing up and breaking the communications?
Steve
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Re: [Asterisk-Users] Asterisk with Primus Talkbroadband
Jon, Does Primus actually use MGCP though? I've heard mix results (though keep in mind I only became interested in all of this earlier today, so I know very little). I checked the specs on my dlink and it says it's SIPs with no mention of MGCP. However everywhere else says Primus is not SIPs. As far as quality, I've had mixed results as well. One of my co-workers says it sounds horrible, but no one else seems to notice anything. Could be the fact that I'm using a cordless phone though. Incoming call quality seems pretty good. Even when downloading at 500K/sec+ it still seems to sound pretty good with only minor choppiness. My best suggestion though, is to get it for a month or so and try it out seeing as it's only $20/month. --- Jon Pounder [EMAIL PROTECTED] wrote: I have asked before and got no answers - I am still not clear as to why there is not an MGCP client as part of asterisk - is it a technical reason, no one else wants it, other ? It is my understanding primus is using mgcp, and therefore is unable to directly interface with asterisk, password or not. I would like to hear your experiences with quality though since I am considering switching to it anyway and run the dlink boxes into my channel bank for now, and figure out the software issues later. Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense. I'm completely new to this. ___ 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ 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] Asterisk with Primus Talkbroadband
Jon, Does Primus actually use MGCP though? I've heard mix results (though keep in mind I only became interested in all of this earlier today, so I know very little). I checked the specs on my dlink and it says it's SIPs with no mention of MGCP. However everywhere else says Primus is not SIPs. What is the exact model of your gateway ? There is an M and S model that are very similar in looks/features except one is sip and one is mgcp (at least last time I looked in detail at what they were supplying) What ports did you open in your firewall for this to work ? that should be another way to tell unless you are wide open. I would try it but I really don't want yet another number, I have numbers in Mississauga and St.Catharines I would like to migrate as long as it is decent quality. As far as quality, I've had mixed results as well. One of my co-workers says it sounds horrible, but no one else seems to notice anything. Could be the fact that I'm using a cordless phone though. Incoming call quality seems pretty good. Even when downloading at 500K/sec+ it still seems to sound pretty good with only minor choppiness. My best suggestion though, is to get it for a month or so and try it out seeing as it's only $20/month. --- Jon Pounder [EMAIL PROTECTED] wrote: I have asked before and got no answers - I am still not clear as to why there is not an MGCP client as part of asterisk - is it a technical reason, no one else wants it, other ? It is my understanding primus is using mgcp, and therefore is unable to directly interface with asterisk, password or not. I would like to hear your experiences with quality though since I am considering switching to it anyway and run the dlink boxes into my channel bank for now, and figure out the software issues later. Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense. I'm completely new to this. ___ 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ 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 Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com ___ 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] Question on echo's for Canadian Asterisk users ...
On Wednesday 08 September 2004 14:46, [EMAIL PROTECTED] wrote: Yeah it looks to be the same setup as mine I am going to try out Mark3 and the Aggressive Suppresor as well. I'm using Mark2 and *no* agressive supressor (which surprised me, I thought I had it in there) -A. ___ 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 Primus Talkbroadband
[EMAIL PROTECTED] wrote: Jon, Does Primus actually use MGCP though? I've heard mix results (though keep in mind I only became interested in all of this earlier today, so I know very little). I checked the specs on my dlink and it says it's SIPs with no mention of MGCP. However everywhere else says Primus is not SIPs. What is the exact model of your gateway ? There is an M and S model that are very similar in looks/features except one is sip and one is mgcp (at least last time I looked in detail at what they were supplying) What ports did you open in your firewall for this to work ? that should be another way to tell unless you are wide open. I would try it but I really don't want yet another number, I have numbers in Mississauga and St.Catharines I would like to migrate as long as it is decent quality. As far as quality, I've had mixed results as well. One of my co-workers says it sounds horrible, but no one else seems to notice anything. Could be the fact that I'm using a cordless phone though. Incoming call quality seems pretty good. Even when downloading at 500K/sec+ it still seems to sound pretty good with only minor choppiness. My best suggestion though, is to get it for a month or so and try it out seeing as it's only $20/month. --- Jon Pounder [EMAIL PROTECTED] wrote: I have asked before and got no answers - I am still not clear as to why there is not an MGCP client as part of asterisk - is it a technical reason, no one else wants it, other ? It is my understanding primus is using mgcp, and therefore is unable to directly interface with asterisk, password or not. I would like to hear your experiences with quality though since I am considering switching to it anyway and run the dlink boxes into my channel bank for now, and figure out the software issues later. Hey, I've checked all over and can't find what I need to know, so I'm posting here. I want to use Asterisk with my Primus VoIP service but it seems I need a username and password to authenticate with at Primus. Has anyone had any experience with this? How did you get it? Is it stored somewhere in the D-Link gateway they gave me? Thanks in advance and sorry if this makes no sense. I'm completely new to this. ___ Asterisk-Users mailing list I did a packet sniff and it is definitely MGCP. I find that the quality hasn't been great. I am looking at moving to something different. It frequently drops calls, but I don't know if it is the NAT device that does it. I would like to find something that is a little bit more Asterisk friendly. I just need to find a provider that can give me DIDs in various BC locations. I have the 1120M/PR model. Have a great day! Geoff ___ 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 Primus Talkbroadband
Jon, Hmm I didn't know about the versions thing. I'll have to get the exact model number off the device when i get home. I never set up any forwarding at all for it though. I simply plugged it into my switch and everything was up and running within a few seconds. Not sure if that's a good sign or not, but probably in the end it will turn out Primus is actually MGCP. Like I said, I haven't played around with this much at all, but is there at least a little support for MGCP in Asterisk? There's a /etc/asterisk/mgcp.conf file, and when you run the Asterisk console it has a bunch of mgcp commands. Or am I mistaken? --- Jon Pounder [EMAIL PROTECTED] wrote: What is the exact model of your gateway ? There is an M and S model that are very similar in looks/features except one is sip and one is mgcp (at least last time I looked in detail at what they were supplying) What ports did you open in your firewall for this to work ? that should be another way to tell unless you are wide open. I would try it but I really don't want yet another number, I have numbers in Mississauga and St.Catharines I would like to migrate as long as it is decent quality. ___ 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 Primus Talkbroadband
Geoff, How frequent are your dropped calls? For a while all my calls would go silent but I realized it was after exactly 60 minutes. It's since been increased to 180. Not sure if this is what you were experiencing. Are there any providers in Canada that offer a similar service to Primus that is more Asterisk friendly? I just need it to have numbers in Toronto, and Montreal if possible. Thanks! --- Geoff Nordli [EMAIL PROTECTED] wrote: I did a packet sniff and it is definitely MGCP. I find that the quality hasn't been great. I am looking at moving to something different. It frequently drops calls, but I don't know if it is the NAT device that does it. I would like to find something that is a little bit more Asterisk friendly. I just need to find a provider that can give me DIDs in various BC locations. I have the 1120M/PR model. Have a great day! Geoff ___ 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] Maximum tollerable lag/jitter for IAX2 w/oji tterbuffer enabled?
On Wed, 8 Sep 2004, Kris Boutilier wrote: I'm leaning in the direction of an IAX2 ACK packet being dropped off the network - I've noticed about 0.5% collisions on the wire (this being a half-duplex network) which seems to be contributing to audible pops and clicks with the jitter buffer disabled. The iax2 debug will show what's happening -- but if an ACK is lost the frame that needed acking will be resent. Steve ___ 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 Primus Talkbroadband
I did a packet sniff and it is definitely MGCP. I find that the quality hasn't been great. I am looking at moving to something different. It frequently drops calls, but I don't know if it is the NAT device that does it. I would like to find something that is a little bit more Asterisk friendly. I just need to find a provider that can give me DIDs in various BC locations. I have the 1120M/PR model. Have a great day! Geoff [EMAIL PROTECTED] wrote: Geoff, How frequent are your dropped calls? For a while all my calls would go silent but I realized it was after exactly 60 minutes. It's since been increased to 180. Not sure if this is what you were experiencing. Are there any providers in Canada that offer a similar service to Primus that is more Asterisk friendly? I just need it to have numbers in Toronto, and Montreal if possible. Thanks! --- Geoff Nordli [EMAIL PROTECTED] wrote: I have been talking with Netvoice.ca. They support IAX, but only have DIDs in the Vancouver area. Also look at companies like TxLink Networks and Vonage.ca. They have DIDs across Canada. You can also join the Asterisk-Biz list and ask what people have available. http://lists.digium.com/mailman/listinfo/asterisk-biz Geoff ___ 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] successful echo cancellation!!! (multitech)
We recently had a customer install that went horribly wrong. Serious echo (pots lines into a cac cb) that, although * did a good job getting rid of alot of it, could not get rid of it all. We tried everything, every canceller, gain setting, etc... combination possible to no avail. Both the vegastream and mediatrix boxes also could not get rid of all of the echo. So, on an off chance, we bought an 8 port fxs/fxo/em gateway made by multitech. The echo cancellation on this device is amazing. There is no trace of the echo and the conversation is still full duplex. And, the box works perfectly with asterisk. Unfortunately, the retail price on these boxes is $3k. Just thought I'd share my experience... -- Joe Antkowiak antkojm1 (at) gmail.com ___ 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] How do I get DIDs for remote areas in Canada
I want the ability to setup DIDs in a variety of different remote locations in Canada. There are various providers that have DIDs in major cities, but none that focus on the smaller cities. The question is how do I actually setup these DIDs? Thanks, Geoff ___ 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] Changed * server to static non-nat IP from nat
Sep 8 15:38:23 WARNING[180235]: chan_sip.c:595 __sip_xmit: sip_xmit of 0x8ec83b4 (len 434) to 147.135.0.129 returned -1: Bad file descriptorSep 8 15:38:23 WARNING[180235]: chan_sip.c:595 __sip_xmit: sip_xmit of 0x8ebb95c (len 434) to 147.135.8.128 returned -1: Bad file descriptorSep 8 15:38:23 WARNING[180235]: chan_sip.c:595 __sip_xmit: sip_xmit of 0x8ec819c (len 434) to 147.135.0.128 returned -1: Bad file descriptorSep 8 15:38:23 WARNING[180235]: chan_sip.c:595 __sip_xmit: sip_xmit of 0x8ed0354 (len 428) to 10.0.2.200 returned -1: Bad file descriptor Anyone know why I would be getting this now. The first three are from BroadVoice and the last one is my old SIP phone which is now a different IP. Did I miss a configuration somewhere when changing from nat to no nat and different subnet? -- Richard Cook [EMAIL PROTECTED] Tel: 705-497-9320- ext 2010 Blank Bkgrd.gif___ 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] 'Hangup' not hanging-up, is this intended behaviour?
Greetings folks;
I have a bit of a conundrum, and I can't tell if Asterisk is doing
something daft, or whether I'm clean missing out why it's doing what it's
doing. So, I have a dialplan that looks a little like this:
[start]
include = dids
include = everythingelse
[dids]
; Test
exten = 8378,1,SetCallerID(3015551212)
exten = 8378,2,Hangup
[everythingelse]
exten = _.,1,AGI,MyScript|${EXTEN} ${ACCOUNTCODE}
The 'Hangup' on 8378 doesn't hang up. Instead it falls through to
'everythingelse' context, and the AGI is executed in the hangup priority.
-- Executing SetCallerID(SIP/3015551212-5acc, 3015551212) in new stack
-- Executing Hangup(SIP/3015551212-5acc, ) in new stack
== Spawn extension (start, 8378, 2) exited non-zero on 'SIP/3015551212-5acc'
-- Executing AGI(SIP/3015551212-5acc, MyScript|h 3015551212) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/MyScript
-- AGI Script MyScript completed, returning 0
If I dial something else (MyScript processes outbound toll/intl calls),
the same as the above happens - the script is again run immediately after
it ends, in the hangup priority. The script is exiting with 0 as a return
code on completion.
When I ask it to Hangup, I expect it to Hangup. I'm guessing that the
_. catch-all is also catching priorities as well as all extensions, which
may or may not be a feature of Asterisk. How do I stop this from
happening?
Thanks all for your help;
JP
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[Asterisk-Users] Re: Avoiding IAX destroy deadlock
On one of my 3 * servers I get this after 2 or 3 IAX2 calls Apr 22 15:54:39 NOTICE[1150495040]: chan_iax2.c:1271 iax2_destroy: Avoiding IAX destroy deadlock And as if that wasn't enough I get a never ending stream of this error flying off the top of the screen. At which point I can no longer make any calls into or out of the box. Any commands issued at the CLI prompt are ignored so I have to do a service asterisk restart now to get it back into service again. Ideas? I am seeing a very similar thing. Ideas? Sep 8 12:17:46 NOTICE[-1137718352]: chan_iax2.c:1271 iax2_destroy: Avoiding IAX destroy deadlock Fedora Core 1, Asterisk CVS-03/30/04-16:59:11. Asterisk is stuck in a select call. Here is a backtrace: (gdb) bt #0 0x0048dc32 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x00579b21 in ___newselect_nocancel () from /lib/tls/libc.so.6 #2 0x08087384 in main (argc=0, argv=0x0) at channel.h:798 I'm not experienced with multi-threaded debugging, so I don't know what any of the other threads were doing. I'm assuming that one of them was responsible for the iax2_destroy call, but I didn't know how to get a backtrace on the thread. I had to restart Asterisk so that phone calls could go through. # ps -C asterisk PID TTY TIME CMD 8356 ?00:10:26 asterisk # strace -p 8356 Process 8356 attached - interrupt to quit select(0, NULL, NULL, NULL, NULL # uname -a Linux server 2.4.22-1.2115.nptl #1 Wed Oct 29 15:42:51 EST 2003 i686 i686 i386 GNU/Linux # grep -c destroy deadlock /var/log/asterisk/messages 848934 # ls -l /var/log/asterisk/messages -rw-r--r-- 1 root root 57236862 Sep 8 12:58 /var/log/asterisk/messages Other log entries from the 7th (surrounding the start of these error messages) There are no other entries from the 6th or 8th. Sep 7 14:06:46 WARNING[-1232168016]: Timeout, but no rule 't' in context 'from-sip' Sep 7 14:07:17 WARNING[-1232168016]: Timeout, but no rule 't' in context 'from-sip' Sep 7 15:03:54 WARNING[-1232168016]: Unable to read password Sep 7 17:53:30 WARNING[-1232168016]: Invalid extension, but no rule 'i' in context 'from-sip' Sep 7 17:53:57 WARNING[-1232168016]: Timeout, but no rule 't' in context 'from-sip' Dustin ___ 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] FXOs
Just recently installed a multitech mvp810 instead of a t100p and cac adit channel bank. Works perfectly, got rid of all echo issues that nothing else had been able to (all the zap echo cancelers, mediatrix gateway, vegastream gateway, etc etc...) ___ 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
