[Asterisk-Users] will digium hardware and asterisk function in asia (korea)?
I am looking at asterisk as a PBX for our import/export business which currently has offices in the US and Korea. Asterisk seems great for our purposes, but I'm somewhat of a telcom newbie and have some questions. Does anybody know anything about the phone system in Korea with respect to whether or not digium hardware and * will function the same as they would in the states? I understand that the voip part is probably location-agnostic, but what about the POTS? Brad Wiemerslage [EMAIL PROTECTED] Seoul, South Korea ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Gogoif with variables acting funny?
Andrew Kohlsmith wrote: On Monday 12 July 2004 18:44, Ed Pringle wrote: $[expr1 operator expr2] Spaces (and lack of spaces) are important. There is no space between the opening [ and expr1, or between expr2 and the closing ]. But you do need spaces separating expr1 from operator, and separating operator from expr2. Any particular reason why it's so picky about spaces, especially between the [] and exprs? Seems like a minor bug to me. -A. I added code to improve the parser, to a degree, a number of weeks ago. It is in CVS right now. Basically, it made it so it didn't care how many spaces were between tokens (as long as there is at least one), or at the beginning or end of the string to be evaluated. It also improved the error messages that are sent to the log (see /var/log/asterisk/messages). And, I made it use double quotes to force a string token... even if the string contains spaces. It's all documented in the asterisk/doc/README.variables. I was very tempted to change it so that it used a lexer-- like lex, perfect hash, etc, etc but just didn't have the time. It'd be a big change. The lexical analysis is real simple. it uses a space, basically, to separate tokens. And that's it! No space? it's all one token. murf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CISCO 7960 VLAN
As far as I know, if your switch doesn't support CDP, you need to configure the VLAN on the phone. It's in Settings - Network Configuration - Option 22 Admin VLAN Id. You will need to unlock the configuration first (method depends on the SIP firmware version you have). -Shaun On Tue, 13 Jul 2004 01:16:03 -0400, Kevin [EMAIL PROTECTED] wrote: I noticed in the Cisco documentation that the access port( the port to hook to a PC) on the 7960 can be configured via CDP with a layer3 Cisco switch. I also see where in the SIP configuration that you can specify the ADMIN VAN. Does anyone know to configure the 7960 access port to use a different VLAN using a non Cisco switch? Thanks, Kevin ___ 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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
Andrew Kohlsmith wrote:
(B
(BI wasn't talking about bandwidth but rather lengthy
(BDial() commands...
(B
(Bexten = s,1,Dial(SIP/someuserSIP/someuserSIP ..
(B
(Bkind of thing... seems awfully unwieldy.
(B
(BThat's why you would stick the members into a global
(Bvariable
(B
(B[globals]
(B
(BDIYCALLGROUP = SIP/111SIP/112SIP113 etc.
(B
(Bthen dial using Dial(${DIYCALLGROUP},...)
(B
(BAlso, you can use the callgroup feature in sip.conf
(B
(B[111]
(B...
(Bcallgroup=1
(Bcallerid="Member 1"12345
(B
(B[112]
(B...
(Bcallgroup=1
(Bcallerid="Member 2"12345
(B
(B[113]
(B...
(Bcallgroup=1
(Bcallerid="Member 3"12345
(B
(Bthen in your dialplan
(B
(Bexten = 12345,1,Dial(SIP/111) ; dialling one member
(Brings them all
(B
(Bthis should call the entire call group. There have been
(Bsome issues with callgroups and SIP some while ago but
(Bthey may have been fixed.
(B
(BIn the event that they haven't been fixed, I suggest once
(Bagain that the bounty would be better spent on fixing
(Bwhatever issues there may still be with callgroups in SIP.
(B
(Brgds
(Bbenjk
(B
(B__
(BDo You Yahoo!?
(Bhttp://bb.yahoo.co.jp/
(B
(B___
(BAsterisk-Users mailing list
(B[EMAIL PROTECTED]
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Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Kannaiyan Natesan wrote: (B (BI hope you clearly understand that everyone here (B**KNOWS** (Bto use alternative software such as SER, what is needed (Bhere is (Bthe same facility in asterisk. (B (BYou have not shown us ANY example yet for which this (Bfacility is *NEEDED*. (B (BYou have only shown us examples for which the facility MAY (Bbe used, all of which have been shown to have OTHER, (Bbetter solutions. (B (BFor call centres you use call queues, for taking workload (Boff admins you use self provisioning, for call groups you (Buse callgroup=1 or a dial string with multiple (Bdestinations, for multi line SIP phones you use multiple (Bextensions. (B (BNone of those problems warrant the use of parallel (Bforking. (B (BYour problem seems to be that you want a facility for its (Bown sake, not because you really need it. That, however is (Bnot good enough a reason to add something to Asterisk. (B (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] segmentation fault on asterisk startup
Hi, I write to this list, because I didn't find anything on the net. I installed asterisk using bristuff-0.0.2 without any errors, but when I start asterisk with asterisk -vvvc I get following error: [codec_ilbc.so] = (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ILBC to SLINR, cost 127 Segmentation fault Removing codec_ilbc.so from /usr/lib/asterisk/modules shows up the next error: [codec_lpc10.so] = (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format LPC10 to SLINR, cost 63 Segmentation fault Ok, just removed this last module works, asterisk is starting without errors anymore, but I wanted to use ILBC codec so it's importan for me. Can anyone help me, getting this to work? I'm running Debian 3.0 (2.4.18-bf2.4) with bristuff-0.0.2 and the zaphfc module loaded. Thanks for any replies. Bye Andreas signature.asc Description: OpenPGP digital signature
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Duane wrote: (B (BWe're running SER and Asterisk on the same system with (BLike2Fone.com and we just stuck Asterisk on a different (Bport then redirect calls as needed, although I doubt it (Bwould (Bbe as difficult as your making out, if you stick SER on (Ban (Balternative port and then just use that to connect your (Bclients to problem solved, in effect the opposite to what (Bwe wanted to achieve... (B (BInteresting. I assume by "redirect calls as needed" you (Bmean passing calls between Asterisk and SER. (B (BIt is unclear to me how you achieve that. (B (BIf Asterisk is directed to speak SIP on port 5061 and SER (Bremains on port 5060, then how do you get Asterisk to talk (Bto SER and vice versa? (B (BWould you care to share this with us? (B (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Kannaiyan Natesan wrote: (B (B I hope you clearly understand that everyone here (B **KNOWS** (B to use alternative software such as SER, what is needed (B here is (B the same facility in asterisk. (B (B You have not shown us ANY example yet for which this (B facility is *NEEDED*. (B (B (BHave you used 5 welcome service in fwd? (BIf not try that. You are invited to join as a volunteer to provide support (Band talk to new people on fwd. (B (BAs I explained to you before we use it for our customer service in call (Bcenter and implemented in many call centres which really makes $. (B (BCan you help me to know how that be achieved with * alone. (B (B-Kannaiyan. (B (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous - Implementation
Based upon the analysis I think we need to modify two things, (B (B1. chan_sip.c (Registrar) (B2. app_dial.c (Dial Command for simultaneous dialling, as of now it (Bsupports simultaneous dialling too) (B (BThe registrar of SIP need to collect the array of registrants and the Dial (Bcommand need to take care of dialling to all possible registrants which I (Bthink should be easier to implement. Anybody thinks will there be any other (Bproblems in handling the same? (B (B-Kannaiyan. (B (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD, DISA DTMF
I've solved. I've putting rfc2833 also on SIP client that connect to first asterisk. Igor On Mon, 2004-07-12 at 22:53, Igor Barsanti wrote: I can dial from an asterisk host to another one via FreeWorldDialup, on the other side DISA service answer to me and i can ear dialtone. But i cannot send DTMF and dial an extension on the DISA enabled asterisk.i've tried rfc2833 and inband...but nothingany tips ??? Thanks, -- Igor Barsanti GPG Public key available at http://pgp.mit.edu http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0xD29D4C21 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
You have not shown us ANY example yet for which this facility is *NEEDED*. Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. When the PBX dials them, all their phones should ring. Asterisk doesn't really bother with *users*, it has a device-centric view of life, universe and propably everything. With Asterisk, the user has to call me each time he wants a new device connected and I have to reconfigure his setup. If I had support for multiple registrations on one [peer] account, the [peer] would become a user account instead of a device. And the user could add as many devices as he wanted (up to a defined limit) without bothering the administrator. I guess that's why a lot of people ask for this function. However, since Asterisk doesn't really bother with a user concept, we really have to teach Asterisk about users. And user groups. Life is much more than hardware, little Asterisk :-) I've been discussing this many times, and so has many other people. I think we need an elegant way of defining users to asterisk so we connect peers, users, agents and mailboxes to a *user* with one set of credentials. If you look into your Asterisk configuration, you will find that there are users and credentials for logging in everywhere. It's not easy to maintain at all. After a lot of discussions on the IRC, I'm convinced that we at some point in time have to add ast_auth - a common infrastructure for handling users and authentication. This is a good topic for the Asterisk Developer's Day at Astricon. Let's bring it up on the agenda - A new user and authentication structure for Asterisk. YALMIATASQ - Yet Another Long Mail in answer to a short question. Hint: I have a new idea for a solution on multiple reg's. Raise the bounty and I might give it a try. ;-) /Olle ___ 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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
Also, you can use the callgroup feature in sip.conf [111] ... callgroup=1 callerid=Member 112345 [112] ... callgroup=1 callerid=Member 212345 [113] ... callgroup=1 callerid=Member 312345 then in your dialplan exten = 12345,1,Dial(SIP/111) ; dialling one member rings them all Seems like a s a weird setup. I can't call them individual that way, can't I? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Ok I'll kick in $25 (just based on your email alone). Is there a formal system for bounty registration? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Tuesday, 13 July 2004 5:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous You have not shown us ANY example yet for which this facility is *NEEDED*. Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. When the PBX dials them, all their phones should ring. Asterisk doesn't really bother with *users*, it has a device-centric view of life, universe and propably everything. With Asterisk, the user has to call me each time he wants a new device connected and I have to reconfigure his setup. If I had support for multiple registrations on one [peer] account, the [peer] would become a user account instead of a device. And the user could add as many devices as he wanted (up to a defined limit) without bothering the administrator. I guess that's why a lot of people ask for this function. However, since Asterisk doesn't really bother with a user concept, we really have to teach Asterisk about users. And user groups. Life is much more than hardware, little Asterisk :-) I've been discussing this many times, and so has many other people. I think we need an elegant way of defining users to asterisk so we connect peers, users, agents and mailboxes to a *user* with one set of credentials. If you look into your Asterisk configuration, you will find that there are users and credentials for logging in everywhere. It's not easy to maintain at all. After a lot of discussions on the IRC, I'm convinced that we at some point in time have to add ast_auth - a common infrastructure for handling users and authentication. This is a good topic for the Asterisk Developer's Day at Astricon. Let's bring it up on the agenda - A new user and authentication structure for Asterisk. YALMIATASQ - Yet Another Long Mail in answer to a short question. Hint: I have a new idea for a solution on multiple reg's. Raise the bounty and I might give it a try. ;-) /Olle ___ 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] Door Phone Question.
I am trying to implement the following. Any idea would help. I have a door phone installed. You know.. Press the button an extension is dialed. right now the extension says a message and then rings the house. If no one answers the person at the door is asked to leave a message. I want the extension to do the following. 1. User presses button. 2. System asks for user to leave a message stating purpose and name after the tone. The message also reminds the caller to press the clear button. 3. The house is rang with caller id stating doorbell. 4. Someone picks up and hears the callers message. The person in the house has the option to 1. replay message, 2. Ring the door to talk. 3. play generic message that no one is around. 5. If no one answers play the no one is around message. -- David Hickman Pots314-865-4752x1 business x31 home FWD 23633 IAXTEL 700-865-4752 AOLIM fsckrmrf ICQ 7059948 Yahoo dhickman THIS IS INSANE! I THOUGHT TECHNOLOGY WAS SUPPOSED TO SIMPLIFY MY LIFE!! ___ 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] Beep during call recording
I want the phone system to play a beep every few seconds to remind the callers that the call is being recorded. Is there any way to do this? thanks dhh -- David Hickman Pots314-865-4752x1 business x31 home FWD 23633 IAXTEL 700-865-4752 AOLIM fsckrmrf ICQ 7059948 Yahoo dhickman THIS IS INSANE! I THOUGHT TECHNOLOGY WAS SUPPOSED TO SIMPLIFY MY LIFE!! ___ 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] X100P ring/off-hook in strange state 6
i have installed asterisk with two X100P cards, everything is working properly but when the channel is answered the following warning appears: WARNING[229391]: chan_zap.c:3073 zt_handle_event: Ring/Off-hook in strange state 6 on channel 1 and this warning is causing some cut off in the sound. i have looked at all previous posts but there is nothing that solved my problem. is there anyone that can help, please thanx ___ 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: Audio filters (was: feature - VM gain adjust?)
Hiya, This is an excellent idea, and is extendable outside of the narrow scope of audio quality improvement. I was playing with this concept a while back, and trying to find programmers for a few ideas I have. I'll air them here, so I can take some credit for being the first clever monkey to publicly talk about integration into Asterisk (or any other VoIP system, as far as I know): - voice disguise/modulation. Think about how many customers you'd get with a module that sounds like they're Mickey Mouse. You think: 'That's really stupid!' but then look at how many Please press 7 for Darth Vader -- I am your father, Luke :-D - voice stress analysis. If you're dumping the audio through a filter, there's no reason you can't simply extract data from it instead of alter the audio path. A one-way background audio carrier tone to the listener might change pitch during stress events. Is there allready some application to do a voice stress analysis? I guess developing something like this from scratch would be very hard... - customized background noise. This is apparently already the rage in Asia somewhere with some cell phone carriers - insertion of background sounds customized to the user's tastes (forest, construction site, bar, office environment, airport, etc.) which can be used for either pleasant diversion or for disguise of location. yeah, this would rock. Honey, i've to stop talkin', the Dentist want's to start drilling. For some Cellphones, this allready exists: http://www.simeda.com/soundercover.html This could also be used to do (MusicDuringCall. Get a call from the army and you play Status Quo (http://www.france-jeunes.net/paroles/index.php?tid=MTkwOTQ=) :-) I have a few more, even, but as is typical, these will remain on the drawing board until someone coughs up some dough to make them happen. No time, no time, no time... Hey, no normal person uses asterisk at home anyway, so there HAS to be some geek out there who also wants this AND can code :-D Bye Andreas _ Watch movie trailers online with the Xtra Broadband Channel http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Hello,
From: Sunrise Ltd [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Date: Tue, 13 Jul 2004 16:31:58 +0900 (JST)
snip
If Asterisk is directed to speak SIP on port 5061 and SER
remains on port 5060, then how do you get Asterisk to talk
to SER and vice versa?
Would you care to share this with us?
It is something like this:
Asterisk extensions.conf:
[globals]
SERADDRESS=XXX.XXX.XXX.XXX:5060
[context]
exten = yourexten,1,Dial(SIP/[EMAIL PROTECTED],20,r)
In ser.cfg:
if (method == INVITE) {
if (uri =~ sip:[EMAIL PROTECTED]){
log(1, Forwarding to Asterisk\n);
rewritehostportt(XXX.XXX.XXX.XXX:5061);
t_relay();
break;
}
}
rgds
benjk
Regards, Girish
_
Earn without investing. http://go.msnserver.com/IN/52048.asp Sell anything
under the sun.
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Re: [Asterisk-Users] Problems with chan-capi
On Mon, 2004-07-12 at 23:53, Andreas Bayer wrote: Hi, i have a debian-system with the asterisk 1:0.9-1 packages and chan-capi 0.3.1-2 installed. My chan-capi seems to be out-of-order. Capi and I4l work in general. I can use isdnlog and capi4hylafax. Using chan_modem_i4l with the sam context work fine too. But no incoming calls are answered by asterisk over chan-capi. If have 2 isdn-card: a passive avm a1 (i4l/hisax) and a active avm b1 (capi). I also tried the 1.0-1 asterisk packages from debian testing. Could you please provide your capi.conf etc. It is hard to give any suggestions, when you just tell it doesn't work, but not supply the configuration. Kind regards, Martin List-Petersen ___ 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] segmentation fault on asterisk startup
Ok, just removed this last module works, asterisk is starting without errors anymore, but I wanted to use ILBC codec so it's importan for me. Can anyone help me, getting this to work? Start off with running ulimit -c unlimited before you start asterisk. Once it crashes, type gdb /path/to/asterisk core From there, enter the following: bt x/5i $eip info registers info threads and quit out. After doing that, you might want to save the output of uname -a cat /proc/cpuinfo and send it to the list. (Note for other people/developers, perhaps something similar to samba's panic action might be useful, which automates a lot of this stuff might come in use.) Hope this helps, Andrew Griffiths ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
I can see the point of the discussion somewhere, but let's take it the other way around (comments though mail): On Tue, 2004-07-13 at 08:53, Olle E. Johansson wrote: You have not shown us ANY example yet for which this facility is *NEEDED*. Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. When the PBX dials them, all their phones should ring. Now .. the problem is, that every hardware phone, every softphone etc. actually might need a different configuration, some IAX, some SIP, some one codec, some other codecs (now that we are talk asterisk). It will get quite problematic to get all solutions under one account without breaking one or the other. Asterisk doesn't really bother with *users*, it has a device-centric view of life, universe and propably everything. With Asterisk, the user has to call me each time he wants a new device connected and I have to reconfigure his setup. Or you provide him with a webinterface, where he has one username and one password. He manages the accounts and settings for each friend,peer,user there and you wouldn't have the work. Kind regards, Martin List-Petersen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Audio filters (was: feature - VM gain adjust?)
- voice stress analysis. If you're dumping the audio through a filter, there's no reason you can't simply extract data from it instead of alter the audio path. A one-way background audio carrier tone to the listener might change pitch during stress events. Is there allready some application to do a voice stress analysis? I guess developing something like this from scratch would be very hard... liarliar.sourceforge.net gives you something, its still in development. When I looked at it a while ago I couldn't get anything useful from it, but I only spent like 5-10 minutes on it. At some stage I was thinking about hooking liarliar up to asterisk to see if the concept would work. A friend of mine raised the point that some codecs won't give you the info you're after most likely over IP, so I never looked further into it. - andrewg ___ 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] caller id problem on incominc call to x100p
hi, when i call asterisk (on x100p) i got this : CLI -- Starting simple switch on 'Zap/7-1' Jul 13 15:03:34 ERROR[311316]: callerid.c:192 callerid_feed: fsk_serie made mylen 0 (-9) Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4735 ss_thread: CallerID feed failed: Success Jul 13 15:03:34 WARNING[311316]: chan_zap.c:4777 ss_thread: CallerID returned with error on channel 'Zap/7-1' but if on the same analog telco line plugin phone i got correct callerid. I have latest cvs asterisk .. (few days old) what can be wrong? brgd, Tomaz ___ 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 to use direcotory from Voicemail
can voicemail be setup to allow a calling user into voicemail to access the the direcotry() ? or can a voicmail subscriber be setup to send(or forward) a voice mail to other users using the same directory() feature? ___ 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] segmentation fault on asterisk startup
[EMAIL PROTECTED] wrote:
Start off with running ulimit -c unlimited before you start asterisk. Once it
crashes, type gdb /path/to/asterisk core
From there, enter the following:
bt
x/5i $eip
info registers
info threads
and quit out. After doing that, you might want to save the output of
uname -a
cat /proc/cpuinfo
and send it to the list.
(Note for other people/developers, perhaps something similar to samba's panic
action might be useful, which automates a lot of this stuff might come in
use.)
Hope this helps,
Andrew Griffiths
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Ok I did.
uname -a gave me this:
Linux chaospbx 2.4.18-bf2.4 #1 Son Apr 14 09:53:28 CEST 2002 i686 unknown
cat /proc/cpuinfo:
processor : 0
vendor_id : CyrixInstead
cpu family : 6
model : 1
model name : 6x86MX 2.5x Core/Bus Clock
stepping: 4
cpu MHz : 166.405
fdiv_bug: no
hlt_bug : no
f00f_bug: no
coma_bug: yes
fpu : yes
fpu_exception : yes
cpuid level : 1
wp : yes
flags : fpu de tsc msr cx8 pge cmov mmx cyrix_arr
bogomips: 331.77
of course I think you wanted the gdb output, I hope that's correct:
(gdb) bt
#0 0x3aeb in ?? ()
#1 0x405e2752 in iLBC_encode (bytes=0x810fda0 ÿ ÿú\017`\022\021¢G\\214,
block=0xb47c, iLBCenc_inst=0x810e868) at iLBC_encode.c:93
#2 0x405e0eea in lintoilbc_frameout (tmp=0x810e868) at codec_ilbc.c:196
#3 0x0805ca2f in calc_cost (t=0x405e9240) at translate.c:238
#4 0x0805ce4a in ast_register_translator (t=0x405e9240) at translate.c:299
#5 0x405e0fef in load_module () at codec_ilbc.c:263
#6 0x080551ce in ast_load_resource (resource_name=0x80defdb
codec_ilbc.so)
at loader.c:312
#7 0x08055636 in load_modules () at loader.c:407
#8 0x08084136 in main (argc=2, argv=0xbe04) at asterisk.c:1485
(gdb) x/5i $eip
0x3aeb: Cannot access memory at address 0x3aeb
(gdb) info registers
eax0xbfffd924 -1073751772
ecx0xbfffd974 -1073751692
edx0x3 3
ebx0x4001e89c 1073866908
esp0xbfffd450 0xbfffd450
ebp0xbfffd99c 0xbfffd99c
esi0x4012819c 1074954652
edi0x40231a9d 1076042397
eip0x3aeb 0x3aeb
eflags 0x10282 66178
cs 0x23 35
ss 0x2b 43
ds 0x2b 43
es 0x2b 43
fs 0x2b 43
gs 0x2b 43
fctrl 0x37f895
fstat 0x122290
ftag 0x 65535
fiseg 0x23 35
fioff 0x405e4895 1079920789
foseg 0x2b 43
fooff 0xbfffd920 -1073751776
fop0x11c284
xmm0 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm1 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm2 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm3 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm4 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm5 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm6 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm7 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
mxcsr 0x1f80 8064
orig_eax 0x -1
(gdb) info threads
* 1 process 8318 0x3aeb in ?? ()
Perhaps it is important to mention, that I got the bad modules again
from a friend. His modules work within my asterisk with no errors.
Very confusing for me. I still hope you can help.
Thanks
signature.asc
Description: OpenPGP digital signature
[Asterisk-Users] zaphfc TE - NT problems
I've got some weird behavior on my HFC-s cards.
asterisk CVS-06/26/04-21:28:35, bristuff 0.02, libpri 20040510, zaptel
20040623
When i pick up my ISDN phone on Zap5-1 (3987) and call the external
number 1901 it will do so, connect me and everything is fine. In the
second, where it tries to attempt the native bridge, the audio will
disappear.
Using another card (Fritz!, chan_capi, same isdn line) works without
problems.
Here is the output from the console:
-- Accepting call from '3987' to 's' on channel 2, span 2
-- Executing DigitTimeout(Zap/5-1, 3) in new stack
-- Set Digit Timeout to 3
-- Executing ResponseTimeout(Zap/5-1, 5) in new stack
-- Set Response Timeout to 5
== CDR updated on Zap/5-1
-- Executing Dial(Zap/5-1, Zap/g1/1901) in new stack
-- Called g1/1901
== D-Channel on span 1 up
== D-Channel on span 1 up
== D-Channel on span 1 up
-- Zap/1-1 is ringing
Jul 13 11:50:15 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI:
received TEI check request for TEI = 73
Jul 13 11:50:17 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI:
received TEI check request for TEI = 73
== D-Channel on span 1 up
== D-Channel on span 1 up
== D-Channel on span 1 up
-- Zap/1-1 answered Zap/5-1
-- Attempting native bridge of Zap/5-1 and Zap/1-1 -- here i loose
all voice
Jul 13 11:50:21 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI:
received TEI check request for TEI = 73
Jul 13 11:50:23 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI:
received TEI check request for TEI = 73
== D-Channel on span 1 up
== D-Channel on span 1 up
== D-Channel on span 1 up
Jul 13 11:50:27 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI:
received TEI check request for TEI = 73
-- Channel 1, span 1 got hangup
Jul 13 11:50:29 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI:
received TEI check request for TEI = 73
-- Channel 1, span 1 got hangup ACK
== D-Channel on span 1 up
== D-Channel on span 1 up
== D-Channel on span 1 up
Jul 13 11:50:35 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI:
received TEI check request for TEI = 73
Jul 13 11:50:36 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI:
received TEI check request for TEI = 73
== D-Channel on span 1 down
== D-Channel on span 1 down
== D-Channel on span 1 down
== D-Channel on span 1 up
Jul 13 11:50:37 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI:
received TEI check request for TEI = 73
Jul 13 11:50:38 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI: !!
Got a UA, but i'm in state 1
Jul 13 11:50:47 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI: !!
Got a UA, but i'm in state 1
-- Channel 2, span 2 got hangup --- here i hang up the phone
-- Hungup 'Zap/1-1'
== Spawn extension (inbound-internal, 1901, 1) exited non-zero on
'Zap/5-1'
-- Hungup 'Zap/5-1'
Jul 13 11:50:57 WARNING[98311]: chan_zap.c:6070 zt_pri_error: PRI:
received TEI check request for TEI = 73
zapata.conf (only the hfc-s cards):
[snip]
switchtype = euroisdn ; HFC-S TE mode
signalling = bri_cpe_ptmp
prilocaldialplan= national
pridialplan = unknown
echocancel = yes
immediate = no
group = 1
context = inbound-zap
nationalprefix = 0
internationalprefix = 00
channel = 1-2
switchtype = euroisdn ; HFC-S NT mode
signalling = bri_net_ptmp
prilocaldialplan= local
overlapdial = no
echocancel = yes
setcallerid = ( ${CALLERIDNUM})
group = 2
immediate = no
context = inbound-internal
channel = 4-5
[snip]
Any suggestions of what is going wrong ?
Kind regards,
Martin List-Petersen
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Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
On 13/07/2004 at 11:48 Martin List-Petersen wrote: I can see the point of the discussion somewhere, but let's take it the other way around (comments though mail): On Tue, 2004-07-13 at 08:53, Olle E. Johansson wrote: You have not shown us ANY example yet for which this facility is *NEEDED*. Well, I have users that get an account on my PBX. I really don't care how many phones they want to use, hardware phones on their desktop or soft phones on their laptop while travelling. It's still a user with one account. When the PBX dials them, all their phones should ring. Now .. the problem is, that every hardware phone, every softphone etc. actually might need a different configuration, some IAX, some SIP, some one codec, some other codecs (now that we are talk asterisk). It will get quite problematic to get all solutions under one account without breaking one or the other. Yes, this is a problem I''d forsee... but ignoring that for one moment :P Imagine that asterisk accepts multiple registrations for a single entry in sip.conf ([myphone]) simply adding each to an internal variable: The first phone registers: WHO_I_DIAL = sip:[EMAIL PROTECTED] then joe comes along and also registers a line on his phone WHO_I_DIAL = sip/[EMAIL PROTECTED]sip/[EMAIL PROTECTED] now when I execute a dial, asterisk internally replaces the occurrence of myphone with the WHO_I_DIAL variable: eg: Dial(SIP/myphone,120) becomes (internally) Dial(WHO_I_DIAL,120) In essence DIAL sees nothing different at all and doesn;t need to be changed because the internal reference SIP/myphone actually = the content of WHO_I_DIAL So what we affectively achieve is: Dial(sip/[EMAIL PROTECTED]sip/[EMAIL PROTECTED],120) Which is what people have been saying everyone should do... but this process becomes automatic, which is a feature that people want. I'm pretty sure you'd do this with an array rather than a string, but I think it explains the theory behind it all. Of course I've ignored the issue with different configs required for different SIP devices (eg DTMFMODE=), but that artistic license ;) I may have explained it badly, so let me know Andy ___ 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 and G729
Try with 'SetGlobalVar' instead of 'SetVar'. Michael. Serge wrote: Dear All, I have problem with new oh323 0.6.3a , I try use var OH323_OUTCODEC, but it don't work. oh323 driver don't want connect to gateway with g729, it's work if I only use in oh323.conf one codec ( g729 ). If I enable 2 or more codecs - always in use other codec: -- Executing SetVar([EMAIL PROTECTED]/1, OH323_OUTCODEC=g729a) in new stack -- Executing Dial([EMAIL PROTECTED]/1, OH323/##|70) in new stack -- H.323 call to # with codec GSM Due Gateway don't support GSM and ulaw, always return: No one is available to answer at this time Many thanks for your help, Regards, Serge. ___ 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] segmentation fault on asterisk startup
On Tue, Jul 13, 2004 at 01:24:18PM +0200, Andreas 'TheChaos' Groll wrote:
[EMAIL PROTECTED] wrote:
Start off with running ulimit -c unlimited before you start asterisk. Once
it crashes, type gdb /path/to/asterisk core
From there, enter the following:
bt
x/5i $eip
info registers
info threads
and quit out. After doing that, you might want to save the output of
uname -a
cat /proc/cpuinfo
and send it to the list.
vendor_id : CyrixInstead
model name : 6x86MX 2.5x Core/Bus Clock
cpu MHz : 166.405
flags : fpu de tsc msr cx8 pge cmov mmx cyrix_arr
bogomips: 331.77
Is anyone else running asterisk with iLBC without problems on cyrix chips?
IIRC, they where meant to be a cheaper version, so initially it made me think
that it might of been gcc emmitting a bad instruction for that cpu.
of course I think you wanted the gdb output, I hope that's correct:
Looks good :)
(gdb) bt
#0 0x3aeb in ?? ()
#1 0x405e2752 in iLBC_encode (bytes=0x810fda0 ? ??\017`\022\021?G\\214,
block=0xb47c, iLBCenc_inst=0x810e868) at iLBC_encode.c:93
#2 0x405e0eea in lintoilbc_frameout (tmp=0x810e868) at codec_ilbc.c:196
#3 0x0805ca2f in calc_cost (t=0x405e9240) at translate.c:238
#4 0x0805ce4a in ast_register_translator (t=0x405e9240) at translate.c:299
#5 0x405e0fef in load_module () at codec_ilbc.c:263
#6 0x080551ce in ast_load_resource (resource_name=0x80defdb
codec_ilbc.so)
at loader.c:312
#7 0x08055636 in load_modules () at loader.c:407
#8 0x08084136 in main (argc=2, argv=0xbe04) at asterisk.c:1485
(gdb) x/5i $eip
0x3aeb: Cannot access memory at address 0x3aeb
Hmmm, looks like saved EIP got overwritten at some stage. I'm not familar with
the translation code, but it might be possible that its buffer was exceeded,
based upon seeing the iLBC_encode passed with a parameter on the stack. I
don't have the code handy at the moment, after I grab it I'll have
a look over it and reply to this message.
(gdb) info registers
eax0xbfffd924 -1073751772
ecx0xbfffd974 -1073751692
edx0x3 3
ebx0x4001e89c 1073866908
esp0xbfffd450 0xbfffd450
ebp0xbfffd99c 0xbfffd99c
esi0x4012819c 1074954652
edi0x40231a9d 1076042397
eip0x3aeb 0x3aeb
eflags 0x10282 66178
cs 0x23 35
ss 0x2b 43
ds 0x2b 43
es 0x2b 43
fs 0x2b 43
gs 0x2b 43
fctrl 0x37f895
fstat 0x122290
ftag 0x 65535
fiseg 0x23 35
fioff 0x405e4895 1079920789
foseg 0x2b 43
fooff 0xbfffd920 -1073751776
fop0x11c284
xmm0 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm1 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm2 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm3 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm4 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm5 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm6 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
xmm7 {f = {0x0, 0x0, 0x0, 0x0}} {f = {-nan(0x7f),
-nan(0x7f), -nan(0x7f), -nan(0x7f)}}
mxcsr 0x1f80 8064
orig_eax 0x -1
(gdb) info threads
* 1 process 8318 0x3aeb in ?? ()
Perhaps it is important to mention, that I got the bad modules again
from a friend. His modules work within my asterisk with no errors.
Very confusing for me. I still hope you can help.
Thanks
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Re: [Asterisk-Users] CID not appearing via X100P
Prior to upgrading my Zaptel sources everything was working fine. I have a X100P connected to my analogue line. The handset port of the X100P is connected to my desk phone's line 2 input. When the analogue line rings I see the CID on my line 2 but not from Asterisk on line 1 via the Cicso ATA. This used to work fine until I upgraded the sources. You might try increasing the rxgain a digit or two to see how that effects CID. ___ 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: Gogoif with variables acting funny?
So far I have tried various forms of the expression including:
exten = t,2,Gotoif,$[${counter} 3]?s|7:h|1
exten = t,2,Gotoif([${counter} 3]?s,7:h,1)
exten = t,2,Gotoif([ ${counter} 3 ]?s,7:h,1)
exten = t,2,Gotoif([ ${counter} 3] ? s,7 : h,1)
With none of the desired results. It always jumps to hangup:
-- Goto (inbound-analog,h,1)
The most interesting result was from the 1st one:
exten = t,2,Gotoif,$[${counter} 3]?s|7:h|1
In the log it showed:
-- Executing SetVar(Zap/99-1, counter=[0+1]) in new stack
-- Executing GotoIf(Zap/99-1, 0?s|7:h|1) in new stack
-- Goto (inbound-analog,h,1)
According to numerous installation guides, I need bison installed to
process expressions within my extensions.conf. I am running RedHat
Enterprise Linus 3.0 which says bison is installed. However it seems I
am not properly processing the expressions, is their a config file or
PATH variable that can be set or is their some other log file that would
show a bison problem within asterisk?
Andrew Kohlsmith wrote:
On Monday 12 July 2004 18:44, Ed Pringle wrote:
$[expr1 operator expr2]
Spaces (and lack of spaces) are important. There is no space
between the
opening [ and expr1, or between expr2 and the closing ]. But you do
need
spaces separating expr1 from operator, and separating operator from
expr2.
Any particular reason why it's so picky about spaces, especially
between the
[] and exprs? Seems like a minor bug to me.
-A.
I added code to improve the parser, to a degree, a number of weeks ago.
It is in CVS right now. Basically, it made it so it didn't care how many
spaces were between tokens (as long as there is at least one), or at the
beginning or end of the string to be evaluated. It also improved the
error messages that are sent to the log (see
/var/log/asterisk/messages). And, I made it use double quotes to force a
string token... even if the string contains spaces.
It's all documented in the asterisk/doc/README.variables.
I was very tempted to change it so that it used a lexer-- like lex,
perfect hash, etc, etc but just didn't have the time. It'd be a big
change. The lexical analysis is real simple. it uses a space, basically,
to separate tokens. And that's it! No space? it's all one token.
murf
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[Asterisk-Users] Local Calls Not Working
Hi I have managed to set up our Asterisk server and can successfully make and receive calls via an external Asterisk server service provider and our IAX.conf file. I can make SIP to SIP calls to a remote machine on a fixed IP. I cannot make SIP to SIP calls from one internal phone behind our NAT firewall to another internal phone behind our NAT firewall. The call is received and the recipient can answer it, but no voice nor echo can be heard. Any ideas would be greatly appreciated. Regards James
[Asterisk-Users] Meridian Option 11c Asterisk Expert Needed
I've tried to do it myself, but my head is now bleeding from hitting it on the wall so much. We need someone who knows asterisk and Meridian PRI cards to help! If required, we will pay for a day's consultancy in order to get this thing working. Or, do I need to scrap my plans to keep the meridian system (60 phones ...) ... Please say no .. :) Please contact me offline (asterisk at dotr dot com) if you want to sell yourselves :) Julian. ___ 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] Integration with a Siemens HiCom 150E / HiPath 3750
Hello again, sorry for the delay in replying; I've been off for some weeks at a customer's offices and couldn't read my email at work... ePyron Felix Deierlein wrote: Hello Martin, how would you like to integrate? PRI (E1) or BRI (ISDN)? Besides of making calls with VoIP from PC to PC, we'd like that our people abroad could dial company internal extensions through Asterisk using a SIP client. On a second approach, the same people abroad could dial the PSTN using the same method... That should not affect your integration with the legacy pbx. Our scenario is: DTAG -- * HICOM PRI | PRI | SIP Seems pretty much similar to what I intend to setup: PSTN --- HiCom -- * (+SIP cloud) PRI S0-Bus Right now, the only free indoor boards provide a S0-Bus (8 ISDN lines), so I thought of using them instead of a PRI board. Some questions about both scenarios, yours and mine: * is it possible to call VoIP from a PC to PSTN and vice versa? * is it possible to call VoIP from PSTN to an internal line? the idea behind this is to have a co-worker somewhere in the world and s/he could ring me on my desk from her/his PC, and vice versa. Please tell me the magical receipt on a step-by-step basis, as I'm not much into this telco world ;) Sorry, that is not that easy because the receipt depends much on the circumstances. What connection do you have between pstn and hicom? It's a PRI. And you should read everything about the leagacy integration, so you will get an idea, what you want to have. Could you please provide some more information? Reading the legacy integration on the *-WiKi page doesn't clear things up too much... You might want to discuss this off the list. I'd post the final conclusions when finished. In that case: Antwort auf Deutsch wäre auch gut ;) Regards, 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] Re: Audio filters (was: feature - VM gain adjust?)
liarliar.sourceforge.net gives you something, its still in development. Hehe, imagine a phone where you see a red LED flashing if the other person lies to you. When you thing about audio-plugins, you should think more into the direction of LADSPA, see http://www.ladspa.org/ Amplifier: http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.5 Compressor: http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.35 There are possible others there as well. ___ 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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
On Tuesday 13 July 2004 03:07, Sunrise Ltd wrote: (B exten = s,1,Dial(SIP/someuserSIP/someuserSIP .. (B (B That's why you would stick the members into a global (B variable (B (BYou global variable is still unwieldy. All you did was move the problem. (B (B Also, you can use the callgroup feature in sip.conf (B (B [111] (B ... (B callgroup=1 (B callerid="Member 1"12345 (B (B [112] (B ... (B callgroup=1 (B callerid="Member 2"12345 (B (BNow *that* is what I was looking for -- so it is possible to group SIP peers (Blike you can Zap channels :-) (B (BThank you. I don't use SIP unless I have to, but I was hoping Asterisk could (Bhandle SIP grouping to help this particular fellow. :-) (B (B-A. (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] segmentation fault on asterisk startup
Hmm, block is allocated near the top of the stack. Ack, I don't like the iLBC code for the quick 3 minutes or so I looked at it, but it wouldn't surprise me if it was overwriting more than it should be on the stack. Well, I'll hand this off to the developers / people who want to spend longer looking over the code (I just happen to be handy with a debugger occasionally). #0 0x3aeb in ?? () #1 0x405e2752 in iLBC_encode (bytes=0x810fda0 ? ??\017`\022\021?G\\214, block=0xb47c, iLBCenc_inst=0x810e868) at iLBC_encode.c:93 #2 0x405e0eea in lintoilbc_frameout (tmp=0x810e868) at codec_ilbc.c:196 #3 0x0805ca2f in calc_cost (t=0x405e9240) at translate.c:238 #4 0x0805ce4a in ast_register_translator (t=0x405e9240) at translate.c:299 #5 0x405e0fef in load_module () at codec_ilbc.c:263 #6 0x080551ce in ast_load_resource (resource_name=0x80defdb codec_ilbc.so) at loader.c:312 #7 0x08055636 in load_modules () at loader.c:407 #8 0x08084136 in main (argc=2, argv=0xbe04) at asterisk.c:1485 (gdb) x/5i $eip 0x3aeb: Cannot access memory at address 0x3aeb ___ 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] HFC-S card and Unable to create channel of type 'Zap'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 hi, i'm new to * I've installed an hfc-s card (DIGI Micro V) with bristuff 0.0.2; when i try to call outside i get: -- Accepting AUTHENTICATED call from 192.168.1.110, requested format = 1024, actual format = 1024 -- Executing Dial([EMAIL PROTECTED]/2, Zap/g1/0123456) in new stack Jul 13 13:42:49 NOTICE[884752]: app_dial.c:559 dial_exec: Unable to create channel of type 'Zap' == Everyone is busy at this time Jul 13 13:43:07 WARNING[163851]: chan_zap.c:6070 zt_pri_error: PRI: Read on 19 failed: Unknown error 500 Jul 13 13:43:07 NOTICE[163851]: chan_zap.c:6976 pri_dchannel: PRI got event: 6 on span 1 - /etc/zaptel.conf loadzone=it defaultzone=it span=1,1,3,ccs,ami bchan=1-2 dchan=3 - ztcfg -v Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) 3 channels configured. - /etc/asterisk/zapata.conf [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode signalling = bri_cpe_ptmp pridialplan = local prilocaldialplan = local echocancel=yes immediate=yes group = 1 context = local channel = 1-2 *CLI zap show channel 1 Channel: 1 File Descriptor: 17 Span: 1 Extension: Context: local Caller ID string: Destroy: 0 Signalling Type: PRI Signalling Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: alaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps unless TDM bridged, currently OFF PRI Flags: Jul 13 14:20:55 WARNING[16384]: chan_zap.c:7351 zap_show_channel: Failed to get conference info on channel 1 Jul 13 14:20:55 WARNING[16384]: chan_zap.c:7357 zap_show_channel: Failed to get confmute info on channel 1 any help will be very apreciated 10x Maurizio - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA89R44Q/49nIJTlwRAtzBAJ9TPn4Hn6WKECiXYFr9Jnf3f0WrnwCePDX+ O1t5ts8wdlOzBU/HyqQpqh4= =Ujbh -END PGP 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
[Asterisk-Users] Asterisk don't listen to my phones
Hello, First, sorry for my english. I'm a french student. I have a problem with asterisk. I use Budgetone SIP phones. When I dial 555 (VoicemailMain), I hear You have 5 new messages, 1- Read your messages, 2- , etc ... ) But when I dial 1 or 2 or everything else, nothing happen. Are they some lines wich do that asterisk listen my phones ? Thanks for your help, have a nice day Thomas DEILLON ___ 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] segmentation fault on asterisk startup
On Tuesday 13 July 2004 08:22, [EMAIL PROTECTED] wrote: Ack, I don't like the iLBC code for the quick 3 minutes or so I looked at it, but it wouldn't surprise me if it was overwriting more than it should be on the stack. Why wouldn't it surprise you? I have a PRI and have 10 or 12 iLBC codecs running during peak times. I don't understand how you can get from I don't like the sound of iLBC to iLBC must be written poorly. -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] IAX2 calls through IAXTEL.com
I created an account at IAXTEL.com to route 1-700-XXX- calls
through. IAXTEL.com gave me a number (example) of 700-555-6226. I have
made the following changes to my:
/etc/asterisk/extensions.conf:
[iaxtel700]
exten =
_81700XXX,1,Dial(IAX2/myusername:[EMAIL PROTECTED]/${EXTEN:1})
exten =
_81800NXX,1,Dial(IAX2/myusername:[EMAIL PROTECTED]/${EXTEN:1})
/etc/asterisk/iax.conf:
[general]
port=5036
bandwidth=high
disallow=all
allow=gsm
tos=0x18
register = myusername:[EMAIL PROTECTED]
[guest]
type=user
context=guest
[iaxtel]
type=peer
context=inbound-analog
auth=rsa
inkeys=iaxtel
[iaxtel-outbound]
type=peer
username=swoolley
secret=gl0bal
host=iaxtel.com
The good news is that dialing 700-XXX- numbers (at Digium) works
great.
I however have two problems:
1) if I dial 800 numbers, like (800)555-1212, I get a bunch of silence
and the following in my log:
-- Starting simple switch on 'Zap/97-1'
-- Executing NoOp(Zap/97-1, ) in new stack
-- Executing Goto(Zap/97-1, intern-post|818005551212|1) in new
stack
-- Goto (intern-post,817005556226,1)
-- Executing Dial(Zap/97-1,
IAX2/myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 69.73.19.178 (format GSM)
-- Format for call is GSM
-- IAX2[iaxtel-outbound]/3 stopped sounds
The call never seems to go through.
2) Not knowing any other way to test, I have simply picked up my
asterisk SIP and analog phones and dialed my own 700 number
(700)555-6226 to which I get a bunch of silence and the following in my
log:
-- Executing NoOp(Zap/97-1, ) in new stack
-- Executing Goto(Zap/97-1, intern-post|817005556226|1) in new
stack
-- Goto (intern-post,817005556226,1)
-- Executing Dial(Zap/97-1,
IAX2/myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 69.73.19.178 (format GSM)
-- Format for call is GSM
-- IAX2[iaxtel-outbound]/2 stopped sounds
-- Hungup 'IAX2[iaxtel-outbound]/2'
But I do get a:
-- Registered to '69.73.19.178', who sees us as 63.143.35.201:4569
When asterisk is starting up so I belive I am registered.
Can I simply not dial my own 700 number from the same asterisk PBX as
a test or do I have some real problem?
--
Steve Woolley
IT Manager
ADS Telecom, Inc.
59 Skyline Drive
Suite 1250
Lake Mary, Florida 32746
Phone: (407)682-6226 x1110
Fax: (407)682-3455
Cell: (321)229-5311
[EMAIL PROTECTED]
www.adstelecom.com
___
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Re: [Asterisk-Users] HFC-S card and Unable to create channel of type 'Zap'
Ciao ! are you connecting a phone or a pbcx to the isdn card ? Hello Maurizio, Tuesday, July 13, 2004, 2:24:24 PM, you wrote: MM -BEGIN PGP SIGNED MESSAGE- MM Hash: SHA1 MM hi, MM i'm new to * MM I've installed an hfc-s card (DIGI Micro V) with bristuff 0.0.2; MM when i try to call outside i get: MM-- Accepting AUTHENTICATED call from 192.168.1.110, MM requested format = 1024, actual format = 1024 MM -- Executing Dial([EMAIL PROTECTED]/2, Zap/g1/0123456) in new stack MM Jul 13 13:42:49 NOTICE[884752]: app_dial.c:559 dial_exec: MM Unable to create channel of type 'Zap' MM == Everyone is busy at this time MM Jul 13 13:43:07 WARNING[163851]: chan_zap.c:6070 MM zt_pri_error: PRI: Read on 19 failed: Unknown error 500 MM Jul 13 13:43:07 NOTICE[163851]: chan_zap.c:6976 pri_dchannel: PRI got event: 6 on span 1 MM - MM /etc/zaptel.conf MM loadzone=it MM defaultzone=it MM span=1,1,3,ccs,ami MM bchan=1-2 MM dchan=3 MM - MM ztcfg -v MM Zaptel Configuration MM == MM SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) MM 3 channels configured. MM - MM /etc/asterisk/zapata.conf MM [channels] MM ; MM ; Default language MM ; MM ;language=en MM ; MM ; Default context MM ; MM ; MM switchtype = euroisdn MM ; p2mp TE mode MM signalling = bri_cpe_ptmp MM pridialplan = local MM prilocaldialplan = local MM echocancel=yes MM immediate=yes MM group = 1 MM context = local channel = 1-2 *CLI zap show channel 1 MM Channel: 1 MM File Descriptor: 17 MM Span: 1 MM Extension: MM Context: local MM Caller ID string: MM Destroy: 0 MM Signalling Type: PRI Signalling MM Owner: None MM Real: None MM Callwait: None MM Threeway: None MM Confno: -1 MM Propagated Conference: -1 MM Real in conference: 0 MM DSP: no MM Relax DTMF: no MM Dialing/CallwaitCAS: 0/0 MM Default law: alaw MM Fax Handled: no MM Pulse phone: no MM Echo Cancellation: 128 taps unless TDM bridged, currently OFF MM PRI Flags: MM Jul 13 14:20:55 WARNING[16384]: chan_zap.c:7351 MM zap_show_channel: Failed to get conference info on channel 1 MM Jul 13 14:20:55 WARNING[16384]: chan_zap.c:7357 MM zap_show_channel: Failed to get confmute info on channel 1 MM any help will be very apreciated MM 10x MM Maurizio -- Best regards, Alessiomailto:[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] HFC-S card and Unable to create channel of type 'Zap'
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Tuesday 13 July 2004 14:41, Alessio Focardi wrote: Ciao ! are you connecting a phone or a pbcx to the isdn card ? simply, i'm connecting this isdn card to an nt1 plus to call outside... - -- Maurizio Marini GSM +39-335-8259739 Work: +39-0721-855285 Fax +39-0721-859609 Home: +39-0721-950396 -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFA898z4Q/49nIJTlwRAtjvAJ4graOK+ODpNyBmrvQiisKF5CVF3wCfbMJR jhsOV93kXX5p8Ygm1SgJDNY= =0oqd -END PGP 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] segmentation fault on asterisk startup
On Tuesday 13 July 2004 08:55, [EMAIL PROTECTED] wrote: You missed my point. I'm talking about how it does data handling with various loops and memcpys etc. I don't care about the sound quality, nor do I care about how well written it is, I'm just making the observation based on my previous experience based on previous auditing of software. My apologies; I read code as codec and though you were making analysis of the code by listening to the codec. Regards, Andrew ___ 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 crashing with no indication why.
On Jul 12, 2004, at 7:37 PM, TC wrote: one to get the status of the queues, and one to get the status of agents. Would one of these commands happen to be show queues or show agents Yep, actually both of those commands. because this guys do a compelete lock on the agents and the queues lists then they lock the individual nodes when it prints those dtls every time it loops to read that info, Is there a better way possibly to retrieve this information? We use it to determine which of our agents are logged in and how many calls we have pending in each queue, aside from reinventing the wheel by creating our own queue app or adding hooks into the existing one making it a pain to upgrade I haven't come up with any way but to poll these constantly. Maybe someone is doing something similar and could give me a tip on the best way to go about this? when you get your next deadlock do this http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging and see if you are in a _pthread_wait_for_restart_signal. look back down the bt for each thread see if you typically have any calls to chan_agent.c- agents_show or app_queue.c-__queues_show I've got my debugging enabled, so I guess I'll just have to wait now for it to happen again. I'm hoping that it does at least show something held up in the bt. Thanks everyone for your suggestions, I'll post as soon as it goes down again with the results of the bt. If there's anything else I should look for I'd be most grateful for the information. Thanks again, --Daniel Daley-- [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] Asterisk crashing with no indication why.
Depending on your dev skills, you could run asterisk in gdb and then look at the status of each thread when the problem occurs. Other than that, try an older version of asterisk PS Please don't post in both lists, it isn't a dev question We're using quite a few of the latest cvs features so I'm going to go ahead and try the deadlock tips on the wiki that everyone has suggested and see where that goes. Sorry about the posting in both lists, I got a little ahead of myself thinking it would eventually turn into a dev issue, I'll wait and see if it goes that direction. Thanks, --Daniel Daley-- [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: SIP simultaneous registry possible workaround (was Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous registry)
On Tue, 2004-07-13 at 03:54, Holger Schurig wrote: Also, you can use the callgroup feature in sip.conf [111] ... callgroup=1 callerid=Member 112345 [112] ... callgroup=1 callerid=Member 212345 [113] ... callgroup=1 callerid=Member 312345 then in your dialplan exten = 12345,1,Dial(SIP/111) ; dialling one member rings them all Seems like a s a weird setup. I can't call them individual that way, can't I? Sure you can. It just depends on you set up the extensions. For example, you could have a section in extensions.conf for the individual SIP extensions, and another section where all the SIP extensions should be rung (e.g. an IVR menu or something). HTH, Kanwar Systems Aligned Inc. www.systemsaligned.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] Digium Cards in Boxes without Power Connectors
Many of the slim compaq/hp boxes don't have DC power cables at all - the PSU plugs right into the mainboard which plugs directly into the daughter- booard that the SCSI hotplug drives plug into. They don't have floppy drives and the cdrom is a laptop-style job which plugs into that SCSI board directly. Ie. No cables. Chris. Gabriel Millerd wrote (on Jul 12): I doubt the backplane is hardwired to the powersupply. You need to see how power is connected to the backplane. On some of our Dells, we have normal drive connectors on the backplane that the power is jumpered off of the motherboard to. You then could get a Y adapter a jump in the middle there. i dont see anything i could put a Y cable onto. there are no normal drives/cdroms to splice ... i believe it would be the same thing on the compaq DL's as well in voip wiki ___ 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 -- == [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] Meridian Option 11c Asterisk Expert Needed
We've successfully integrated with an Option 61c, but it was painful. We've set up both ends to emulate a 5eSS switch. The Asterisk is using pri_net (meaning the Nortel is pri_cpe (Client Side)). Unfortunately, in this configuration the Nortel thinks that this trunk is connected to an external phone company, so it always sends it's external Caller ID to it. That means that when someone on the Nortel calls someone on the Asterisk, you will always see the external caller id, not the actual extension from which the call originated. Our company uses Qwest to administer the Nortel, so it was the Qwest technicians who actually installed the card, set up the trunk and established the dial plan. We also found that we had to buy a special software option call Custom Dial Plan (CDP)which cost an extra $6,795 including installation. With CDP installed on the Nortel, they were able to create a dial-plan where extensions from 4000 to 4999 were sent down the trunk that's connected to the Asterisk. Asterisk then routes them accordingly. Asterisk has a dial-plan where all extensions from 2000 to 2999 are sent back to the Nortel. In that case, caller id works as it should for both name and number. Asterisk is also configured to send toll calls through the Nortel and that works correctly. So, that's the summary of what we've been able to accomplish. I can provide you with the config files on the Asterisk, but you'll need a Nortel tech for the rest; I have neither the ability nor the access to make those types of changes to a Nortel system. I hope this helps. Joe [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Tuesday, July 13, 2004 7:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed I've tried to do it myself, but my head is now bleeding from hitting it on the wall so much. We need someone who knows asterisk and Meridian PRI cards to help! If required, we will pay for a day's consultancy in order to get this thing working. Or, do I need to scrap my plans to keep the meridian system (60 phones ...) ... Please say no .. :) Please contact me offline (asterisk at dotr dot com) if you want to sell yourselves :) Julian. ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 ___ 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
[Asterisk-Users] Possible Asterisk Notify Bug
I noticed when my Cisco device sends a SUBSCRIBE message to Asterisk for voice mail subscription. The Asterisk server will send the wrong call ID back. Thus, the Cisco sends a 481 back to the Asterisk. I believe the below section in RFC 3265 is relevant: 3.3.4 NOTIFY requests are matched to such SUBSCRIBE requests if they contain the same Call-ID, a To header tag parameter which matches the From header tag parameter of the SUBSCRIBE, and the same Event header field. Rules for comparisons of the Event headers are described in section 7.2.1. If a matching NOTIFY request contains a Subscription-State of active or pending, it creates a new subscription and a new dialog (unless they have already been created by a matching response, as described above). Below is a portion of the trap I obtained from the Asterisk Server. A complete trap can be found at http://www.pasewaldt.com/notify.html Sip read: SUBSCRIBE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bKFC2 From: 2486 sip:[EMAIL PROTECTED];tag=6C30-149 To: sip:[EMAIL PROTECTED] Date: Call-ID: 2C1ED0F6-2BDE11D6-80048294-A080CC2F CSeq: 101 SUBSCRIBE Timestamp: 1089723760 Contact: sip:[EMAIL PROTECTED]:5060 Event: message-summary Expires: 600 Accept: application/simple-message-summary Content-Length: 0 13 headers, 0 lines ^Dasterick*CLI Using latest SUBSCRIBE request as basis request Sending to 192.168.0.1 : 5060 (non-NAT) Looking for 2486 in voice-mail Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK19c0b453 From: asterisk sip:[EMAIL PROTECTED];tag=as288443e6 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 37 Messages-Waiting: yes Voicemail: 7/0 (no NAT) to 192.168.0 Kurt __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ 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 don't listen to my phones
Try configuring your Grandstream to send DTMF via SIP INFO instead of in-audio. -Seth On Tue, 2004-07-13 at 08:33, thomas DEILLON wrote: Hello, First, sorry for my english. I'm a french student. I have a problem with asterisk. I use Budgetone SIP phones. When I dial 555 (VoicemailMain), I hear You have 5 new messages, 1- Read your messages, 2- , etc ... ) But when I dial 1 or 2 or everything else, nothing happen. Are they some lines wich do that asterisk listen my phones ? Thanks for your help, have a nice day Thomas DEILLON ___ 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 -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] Cann't load oh323 0.6.3a
Hi, After a whole day of work, I finally complied oh323 0.6.3a successfully. But when I started asterisk, it cann't load oh323. Following is the error: [format_jpeg.so] = (JPEG (Joint Picture Experts Group) Image Format) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [cdr_csv.so] = (Comma Separated Values CDR Backend) [chan_oh323.so]Jul 13 09:43:45 WARNING[1074460416]: loader.c:240 ast_load_resource: liboh323wrap.so: cannot open shared object file: No such file or directory Jul 13 09:43:45 WARNING[1074460416]: loader.c:408 load_modules: Loading module chan_oh323.so failed! The version of pwlib is 1.6.6, and the version of openh323 is 1.13.5. OS is Redhat 9. which file does it want to open? could you have any idea of this problem? By the way, how to uninstall the oh323? because now I cann't start the asterisk. Thanks a lot. Rui __ Post your free ad now! http://personals.yahoo.ca ___ 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] Meridian Option 11c Asterisk Expert Needed
The config files would be great, thanks ! I'll let you know how I get on :) Julian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: 13 July 2004 15:16 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed We've successfully integrated with an Option 61c, but it was painful. We've set up both ends to emulate a 5eSS switch. The Asterisk is using pri_net (meaning the Nortel is pri_cpe (Client Side)). Unfortunately, in this configuration the Nortel thinks that this trunk is connected to an external phone company, so it always sends it's external Caller ID to it. That means that when someone on the Nortel calls someone on the Asterisk, you will always see the external caller id, not the actual extension from which the call originated. Our company uses Qwest to administer the Nortel, so it was the Qwest technicians who actually installed the card, set up the trunk and established the dial plan. We also found that we had to buy a special software option call Custom Dial Plan (CDP)which cost an extra $6,795 including installation. With CDP installed on the Nortel, they were able to create a dial-plan where extensions from 4000 to 4999 were sent down the trunk that's connected to the Asterisk. Asterisk then routes them accordingly. Asterisk has a dial-plan where all extensions from 2000 to 2999 are sent back to the Nortel. In that case, caller id works as it should for both name and number. Asterisk is also configured to send toll calls through the Nortel and that works correctly. So, that's the summary of what we've been able to accomplish. I can provide you with the config files on the Asterisk, but you'll need a Nortel tech for the rest; I have neither the ability nor the access to make those types of changes to a Nortel system. I hope this helps. Joe [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Tuesday, July 13, 2004 7:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed I've tried to do it myself, but my head is now bleeding from hitting it on the wall so much. We need someone who knows asterisk and Meridian PRI cards to help! If required, we will pay for a day's consultancy in order to get this thing working. Or, do I need to scrap my plans to keep the meridian system (60 phones ...) ... Please say no .. :) Please contact me offline (asterisk at dotr dot com) if you want to sell yourselves :) Julian. ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 ___ 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 ___ 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] Cann't load oh323 0.6.3a
Hello ruixun, Tuesday, July 13, 2004, 6:26:53 PM, you wrote: [format_jpeg.so] = (JPEG (Joint Picture Experts rw Group) Image Format) rw == Registered format 'jpg' (JPEG (Joint Picture rw Experts Group)) rw [cdr_csv.so] = (Comma Separated Values CDR Backend) rw [chan_oh323.so]Jul 13 09:43:45 WARNING[1074460416]: rw loader.c:240 rw ast_load_resource: liboh323wrap.so: cannot open shared rw object file: No rw such file or directory rw Jul 13 09:43:45 WARNING[1074460416]: loader.c:408 rw load_modules: Loading rw module chan_oh323.so failed! Have you executed ldconfig after installing oh323? By default liboh323wrap.so is located in /usr/local/lib and you must add this path to /etc/ld.so.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] Oz ISDN
On Tue, 2004-07-13 at 12:28, Kimble Young wrote: David, If you go the analogue route: * You'll get poor audio compared to ISDN which is crystal. * Each number will act like a seperate line unlike with an ISDN card where you can receive two calls simultaneously on the same line. Actually, you can configure the NT1+II so that it will behave differently. (ie, you can set it so that calls to each number will 'prefer' a specific port, but if that port is in use will use the other port. * You'll lose cool ISDN features like call deflection. Dunno if call deflection even works here, haven't actually got around to trying it yet. * It won't be as reliable (speculation). Well, as with any comparison between analogue/digital, the digital is definitely preferred. You *know* when the other side answers/hangs up/etc. This gives you accurate cdr (billing) information as well as call progress. Stops you from causing the line to be busy long after the other party has hung up. etc... * It'll probably cost just as much for two analogue cards as a fritz card. Probably, I'm not sure, but the frustration factor might in the long run... On the positive side you won't have to go through a lot of frustration getting the fritz working. Dunno about this, I got a fritz card about 2 weeks ago for a customer's pbx. Plugged the card in, followed the instructions available on the wiki, and basically it just worked. Perhaps if I knew less about linux, it might have been harder, but from memory, I didn't do anything specially fancy... If I was doing this for myself, at home, I would definitely use the fritz card. Also, AFAIK, there are no aca approved (old austel tick) fxo cards. In either case, I would suggest you discuss your options with the folks at www.atp.org.au, I've found them to be quite helpful, and definitely quite knowledgeable Regards, Adam ___ 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] Local Call Problems
Hi Further to my previous email... I have a Xten software phone connecting to a Grandstream 100 hardware phone. My first problem is that voice transmits in one direction only. Secondly, this only works if the codecs on both are identical. If the Xten uses GSM and the Grandstream uses ULAW then the phones connect, but no voice can be hears in either direction. I assumed (possibly wrongly) that Asterisk did the appropriate codec translation? Regards James
[Asterisk-Users] chan_oh323
Hello, has anybody managed to register with two gatekeepers using chan_oh323? Lars ___ 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] Dial Fail - Send Email
Hello, I have an asterisk implementation that is running for the last 2 months. Now the customer wants to be able to get an email everytime a dial command fails...i.e when either no one picks up, its busy or the link to the end user device is down. Actually, this is a small call centre type of installation. * is located in Singapore and the end points (i.e agents) are located in India. And the reason he wants this email (not voicemail notification to email, just a notification that the call did not get thru) is because the link between * and the agents may be down.mainly due to the internet connectivity issues. Does anyone have such an app? Also, this app should send multiple email addresses the email that the link may be down. If there isn't such an app, can someone develop it for me? I will be willing to pay a small price for it. Thanks San
RE: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed
Zapata.conf: [channels] context=default switchtype=5ess signalling=pri_net group=1 channel = 1-23 usecallerid=yes hidecallerid=no echocancel=yes echocancelwhenbridged=yes echotraining=yes Zaptel.conf: span=1,2,0,esf,b8zs,yellow bchan=1-23 fcshdlc=24 loadzone = us defaultzone=us That's all there is to it. When it's running, you can access those trunks as 'Zap/1-1', 'Zap/1-2', 'Zap/1-3', etc. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Tuesday, July 13, 2004 9:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed The config files would be great, thanks ! I'll let you know how I get on :) Julian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: 13 July 2004 15:16 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed We've successfully integrated with an Option 61c, but it was painful. We've set up both ends to emulate a 5eSS switch. The Asterisk is using pri_net (meaning the Nortel is pri_cpe (Client Side)). Unfortunately, in this configuration the Nortel thinks that this trunk is connected to an external phone company, so it always sends it's external Caller ID to it. That means that when someone on the Nortel calls someone on the Asterisk, you will always see the external caller id, not the actual extension from which the call originated. Our company uses Qwest to administer the Nortel, so it was the Qwest technicians who actually installed the card, set up the trunk and established the dial plan. We also found that we had to buy a special software option call Custom Dial Plan (CDP)which cost an extra $6,795 including installation. With CDP installed on the Nortel, they were able to create a dial-plan where extensions from 4000 to 4999 were sent down the trunk that's connected to the Asterisk. Asterisk then routes them accordingly. Asterisk has a dial-plan where all extensions from 2000 to 2999 are sent back to the Nortel. In that case, caller id works as it should for both name and number. Asterisk is also configured to send toll calls through the Nortel and that works correctly. So, that's the summary of what we've been able to accomplish. I can provide you with the config files on the Asterisk, but you'll need a Nortel tech for the rest; I have neither the ability nor the access to make those types of changes to a Nortel system. I hope this helps. Joe [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Tuesday, July 13, 2004 7:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed I've tried to do it myself, but my head is now bleeding from hitting it on the wall so much. We need someone who knows asterisk and Meridian PRI cards to help! If required, we will pay for a day's consultancy in order to get this thing working. Or, do I need to scrap my plans to keep the meridian system (60 phones ...) ... Please say no .. :) Please contact me offline (asterisk at dotr dot com) if you want to sell yourselves :) Julian. ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 ___ 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 ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 ___ 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] Digium Cards in Boxes without Power Connectors
- Original Message - From: Gabriel Millerd [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 12, 2004 2:06 PM Subject: [Asterisk-Users] Digium Cards in Boxes without Power Connectors I noticed on the wiki that some of the production hardware (compaq) doesnt have a power connector to my knowledge. I have a compaq c6400 that I would like to use for Asterisk. However all the drives are hot swap and the dual redundant power supply bay are not something i really feel like soldering wiring to. How are people getting around this? Is there a magic 'fan card' that has a power out that people are using? This may work for you. http://www.thermaltake.com/products/subzero/subzero4g.htm J.Christian Hoffmeyer Asterisk Solutions Group, Inc. Huntsville, AL (o)256.705.0265 (c)256.655.0321 (fax) 256.705.0280 (tf)877.ASGI.4.ME (iax) 700.ASGI.4.ME Ask me about Asterisk ___ 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] Oz ISDN
On Wed, 2004-07-14 at 00:42, Adam Goryachev wrote: In either case, I would suggest you discuss your options with the folks at www.atp.org.au, I've found them to be quite helpful, and definitely quite knowledgeable PS, most digium resellers seem to follow the standard digium policy of offering 1 hour of post sales installation support. So if you went the ISDN route, then they would most likely be able to assist you if you got into trouble. The above people seem to offer the same. Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CISCO 7960 VLAN
Shaun, Thanks for the feedback. I am aware of the ADMIN VLAN setting and mentioned it below in my original post. I was referring to the access port (The additional 10/100 port). I wanted to change the access port as it handles tagged or untagged VLAN frames. Kevin -Original Message- From: Shaun Ewing [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 13, 2004 2:18 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] CISCO 7960 VLAN As far as I know, if your switch doesn't support CDP, you need to configure the VLAN on the phone. It's in Settings - Network Configuration - Option 22 Admin VLAN Id. You will need to unlock the configuration first (method depends on the SIP firmware version you have). -Shaun On Tue, 13 Jul 2004 01:16:03 -0400, Kevin [EMAIL PROTECTED] wrote: I noticed in the Cisco documentation that the access port( the port to hook to a PC) on the 7960 can be configured via CDP with a layer3 Cisco switch. I also see where in the SIP configuration that you can specify the ADMIN VAN. Does anyone know to configure the 7960 access port to use a different VLAN using a non Cisco switch? Thanks, Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help Needed in configuring Cisco 7940
I bought a Cisco 7940, I need to configure it for Asterisk. I checked the wiki pages. Followed the link to Cisco web page. Tried to download the image for SIP. It wo'nt allow me even though I registered for the CCO Valet. Is the image available anywhere else? I saw some of the messages in the mailing list that it supposed to be fairly simple. I would highly appreciate if somone could post the step by step configuration process in detail. Thanks in advance.. __ Do you Yahoo!? Yahoo! Mail - 50x more storage than other providers! http://promotions.yahoo.com/new_mail ___ 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 debugging tools
But what exactly are you trying to debug ? Specifically, I want to determine when the DTMF tones are being sent over the channel. I'm connected to a NACT switch using cas, there is a delay of 5-8 seconds from when asterisk begins the Dial command to the time when the NACT switch connects the call. This behavior does not occur with other equipment connected to the NACT switch, so I'm trying to narrow down the problem. * you could issue set verbose 10 on the asterisk CLI I've tried set verbose 10 as you suggested, followed by a debug Zap/1-1 as soon as the call is attempted. I get the following message repeatedly until the call is connected: [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] Any ideas or directions are appreciated. -g On Mon, 2004-07-12 at 19:14, C. Maj wrote: On Mon, 12 Jul 2004, Glen Hinkle waxed: Are there any debugging tools for the digium zaptel cards that would report the activity on the line, such as DTMF and/or connection protocol? * zttool is in the zaptel source directory * you could issue set verbose 10 on the asterisk CLI * you could issue pri debug span x on the asterisk CLI Also, try getting a PRI trace from your telco. But what exactly are you trying to debug ? --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] chan_oh323
Hello, I have been trying for a while to make the oh323 channel working but i didnt manage, i have everything compiled correctly but asterisk find somethign like an "undefined symbol" when it loads the oh323 module... i dont know if u have seen this before, I am deseperate to find the solution , i am involved in a very important project and i am out of time :( I would be very grateful if you can help me... Best Regards, soumayaLars Degenhardt [EMAIL PROTECTED] wrote: Hello,has anybody managed to register with two gatekeepers usingchan_oh323?Lars___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 Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !
Re: [Asterisk-Users] Sort of OT: Recommended USB handset for use with iaxComm?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Nate Carlson wrote: | On Mon, 12 Jul 2004, Brian Weaver wrote: | |Tell him to spend $70 on a one port ATA like the Sipura-1000 , a USB |headset will run $40-60. For a little more money, you'll have an |external box that is not leashing you to a computer or one location. | | | Actually, he'd like the USB handset over the ATA - he's not sure what type | of 'net access he's going to have at places, and doesn't want to have to | deal with provided an ethernet connection to the ATA. I have been successfully using the Plantronics DSP-400 USB headset with both gnophone and iaxComm. gnophone audio is beautiful and is apparently a very good client for recovering from stray bit loss, etc. However, it is no longer supported, even in Asterisk out-of-the-box. You have to compile IAX version 1 support back in if you want to use gnophone. iaxComm is still, in my opinion, quite rough around the edges. The audio quality on transmission to another end is very scratchy and poppy. I'm not sure if this is a function of CPU power or the audio system that iaxComm uses. I've been attempting to use it on an iPAQ device running GPE. I sucessfully compiled it and have it running, but the audio is especially terrible on that device (H3670, 206MHz ARM, 64MiB RAM). I was able to apply the patch that ziaxPhone has for the iax2-parser not being word aligned, and that helped the audio tremendously. But it's still very poppy, echoy, feedbacky, and scratchy. Sort of like lurching through the audio stream. I've learned that if I turn the micrphone down to almost nothing it works a lot better for awhile. After awhile, though (30 to 45 sec.), the audio output almost fades to nothing on the iPAQ device. Anyway. You might take a look at the USB device available from www.virbiage.com. It might work with more than just their FireFly client, especially if it shows up as an audio device under Linux. - -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Debian - http://enigmail.mozdev.org iD8DBQFA9ABmuYsUrHkpYtARAjrXAJ9nPISN+bJ6CCEXiTaIp6OmE8BIQgCdFKwY XwmtVPPDeZSupONLh/yEhjM= =cLr2 -END PGP SIGNATURE- -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ 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] Local Call Problems
Set canreinvite=no in sip.conf to force the RTP voice traffic to pass through asterisk so it can do the transcoding. -Seth On Tue, 2004-07-13 at 10:46, James Dutton wrote: Hi Further to my previous email... I have a Xten software phone connecting to a Grandstream 100 hardware phone. My first problem is that voice transmits in one direction only. Secondly, this only works if the codecs on both are identical. If the Xten uses GSM and the Grandstream uses ULAW then the phones connect, but no voice can be hears in either direction. I assumed (possibly wrongly) that Asterisk did the appropriate codec translation? Regards James -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Audio filters (was: feature - VM gain adjust?)
On Tue, 2004-07-13 at 07:11, Holger Schurig wrote: liarliar.sourceforge.net gives you something, its still in development. Hehe, imagine a phone where you see a red LED flashing if the other person lies to you. When you thing about audio-plugins, you should think more into the direction of LADSPA, see http://www.ladspa.org/ Amplifier: http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.5 Compressor: http://plugin.org.uk/ladspa-swh/docs/ladspa-swh.html#tth_sEc2.35 There are possible others there as well. While that would get quite a bit of plugins right away. It would need to be looked over carefully to make sure every plugin being used and the API is able to handle multithreading. Add to that, are the plugins fast enough to not add much latency to the audio stream. -- Steven Critchfield [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] Applications of TDMoE
On Tue, 2004-07-13 at 06:30, luan wrote: Hi All, Please bear my ignorance but what is TDMoE used for? Illustrations with practical applications, scenarios or set ups will be most appreciated. Start here and then ask a real question. http://www.google.com/search?hl=enq=tdmoe+site%3Alists.digium.com -- Steven Critchfield [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] Asterisk System Proposal
Hello, Earlier last week, I had posted a requirement for Asterisk boxes to various consultants, vendors. Thanks to all of them who responded. After screening through some of the quesitons, we realized that there are too many variables to a working Asterisk system and a remote vendor may not be the best way to go. So it would be best for us to discuss our requirement with someone who is local to San-Jose, Bay Area. I am posting our requirements again on this mailing list hoping someone local can respond to it. Criteria: 1. You are local to Bay Area (CA). 2. You have installed Asterisk systems. 3. You have the confidence to install configure following 3 systems. If you meet the above criteria, please send an email to helloritesh at gmail dot com so that we can exchange phone numbers for a direct conversation. Here is a brief detail of required system. (1) Asterisk PBX : USA * T1 interface to PSTN * 2 Analog PSTN interface of emergency calling (911) * Upto 100 SIP interfaces (including voice mail) * 2U Chassis (2) Asterisk PBX : ASIA * 8 PSTN (Analog) lines * Upto 50 SIP interfaces (including voice mail) * 2U Chassis * You don't need to travel. We can test the system locally and the system will be shipped to ASIA office. (3) Asterisk PBX : Europe * 8 PSTN (Analog) lines * Upto 50 SIP interfaces (including voice mail) * 2U Chassis * You don't need to travel. We can test the system locally and the system will be shipped to ASIA office. Requirements: * Voice-mail back-up * Single Dialplan across offices * GUI for simple maintenance (e.g. adding an extension) * Remote logging to update dialplan using GUI * Firewall pass through for voice traffic (e.g. IAX) * Maintenance contract terms and cost * Detail system warranty information * solution in two to three weeks Initial dialplan will be provided to you and the systems are expected to be pre-configured with the dial-plan. Our IT is in a process of setting up the infrastructure (mail server, file server, firewalls etc.) and the systems need to be deployed in 2 to 3 weeks so please respond at your earliest convenience. Some of the questions we have been asked: Q: Any idea for the call-volume expected out of your US-PBX? A: We do not have a call volume yet. We will probably start with a pri or mutliple analogs. This will probably carry us for at least the first couple of months. Q: Assuming that IAX would be used for inter-site communication: What are the internet access speeds at remote sites? Will this connection/access device before the PBX have QoS/traffic shaping abilities as well as VoIP support (are you willing to open port 4569 for IAX on your firewall, etc) A: We do not yet have the exact speeds on connectivity for the remote sites, some are being reprovisioned. We should not need to open the firewall, as the remote sites should have full time vpn connectivity. Q: Voice-mail back-up: Is disk based backup enough? Bandwidth to copy VM files from US/Europe? A: Ideally we will have fault tolerence in the US, we should be able to do a flat file copy from Europe to the US. Q: Will you have tape drives or other network-backup facilities (or should we provide them) A: We will have tape drives. Thanks a lot for your time and I look forward to your responses. ___ 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] G729A and GSM - newbie question
Hello, When I'm trying to play standard sound files from Asterisk using G729A codec with OH323 channel I get this message: channel.c:1650 ast_set_write_format: Unable to find a path from GSM to G729A It seems that this files must be in G729 format? How can I convert this files to G729? ... or am I wrong? -- wbr, Oleg ___ 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] Digium Cards in Boxes without Power Connectors
On Tue, 2004-07-13 at 09:02, Chris Luke wrote: Many of the slim compaq/hp boxes don't have DC power cables at all - the PSU plugs right into the mainboard which plugs directly into the daughter- booard that the SCSI hotplug drives plug into. They don't have floppy drives and the cdrom is a laptop-style job which plugs into that SCSI board directly. Ie. No cables. So you either can't use that machine or you deal with trying to pull the power through the PCI bus. Any Fan card that you put in is going to just pull the power out of the same PCI bus that the TDM card does. Gabriel Millerd wrote (on Jul 12): I doubt the backplane is hardwired to the powersupply. You need to see how power is connected to the backplane. On some of our Dells, we have normal drive connectors on the backplane that the power is jumpered off of the motherboard to. You then could get a Y adapter a jump in the middle there. i dont see anything i could put a Y cable onto. there are no normal drives/cdroms to splice ... i believe it would be the same thing on the compaq DL's as well in voip wiki ___ 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 -- Steven Critchfield [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] zaphfc does not indicate congestion!?
Hi there, I am using bri-stuff.0.0.2 and maybe I misunderstood something but my HFC card is in bri_cpe_ptmp mode and gets routed about 80 MSNs. Some of them are not intended to be used by asterisk but every incoming call is accepted even if the default extension leads to Congestion: -- Accepting call from '28715' to '27849' on channel 1, span 2 -- Executing Dial(Zap/4-1, SIP/27849) in new stack == Everyone is busy at this time -- Executing Congestion(Zap/4-1, ) in new stack What am I doing wrong? How can I accept only calls to those terminals that are currently available and ready to answer a new call? Deti ___ 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: One way audio... continuously and randomly
:( Just getting silence Is this mailing list alive at all? Vasyl Rublyov wrote: All, I seen already threads about one way audio... but never seen anyone answered completely on it. There is a problem, what we are getting, even with stable-1, CVS updates in May, June as well as last Saturday (Jul 10, 2004) [T1/PRI PSTN] == [Lucent Legend PBX] == [T1/PRI] == [T100P Asterisk IAX2] == [T1 Internet (ISP Verizon = QWest) connected thru T100P interfaces (before it was NetOpia T1 router but the same problem existed)] === [ADSL Internet (ISP: UTEL/Ukraine)] === [IAX2: Asterisk with TDM400 cards] === [Analog phones SIP phones (Cisco 79xx Polycom IP500] Calling from here and thru [T1/PRI PSTN] to final phones, analog or just sip phones, keep dropping calls, but __ALMOST ALWAYS__ called party does not hear when calling party hear well. We tried different settings for IAX - with and without trunking. I see the traffic goes both ways and counters on the trunks/channels are increasing even when no audio in the phone. Digium G729 codec is in used, the same problem was exiting when tested with iLBC GSM codecs, but sounds like DID NOT exist with G711 codec (ULAW) PLEASE HELP At least where should I start look? Thank you in advice Vasyl ___ 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 -- Thanks and regards, Vasyl Rublyov ___ 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] unclean hangups can I turn off hook flash?
I'm having problems with unclean hangups (being read as a flash instead of a hangup?). Can I turn off hook flash recognition in asterisk, but still have the flash button on the analog phone operational? Could I use these settings in zapata.conf to fix my problem? *prewink*: Sets the pre-wink timing. *preflash*: Sets the pre-flash timing. *wink*: Sets the wink timing. *rxwink*: Sets the receive wink timing. *rxflash*: Sets the receive flash timing. *flash*: Sets the flash timing. *start*: Sets the start timing. *debounce*: Sets the debounce timing. The debounce settings in the Asterisk configuration affects how Asterisk handles hookswitch transitions on its FXO/FXS interfaces. -- ___ 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] IAX2 calls through IAXTEL.com
I've been getting the same issue with toll free numbers over IAXTEL for
the last 4-5 days. I contacted Digium support (IAXTEL's website says to)
on July 9th, and all I got back was We will look in to it.
I haven't heard anything since.
On Tue, 2004-07-13 at 08:38 -0400, Steve Woolley wrote:
I created an account at IAXTEL.com to route 1-700-XXX- calls
through. IAXTEL.com gave me a number (example) of 700-555-6226. I have
made the following changes to my:
/etc/asterisk/extensions.conf:
[iaxtel700]
exten =
_81700XXX,1,Dial(IAX2/myusername:[EMAIL PROTECTED]/${EXTEN:1})
exten =
_81800NXX,1,Dial(IAX2/myusername:[EMAIL PROTECTED]/${EXTEN:1})
/etc/asterisk/iax.conf:
[general]
port=5036
bandwidth=high
disallow=all
allow=gsm
tos=0x18
register = myusername:[EMAIL PROTECTED]
[guest]
type=user
context=guest
[iaxtel]
type=peer
context=inbound-analog
auth=rsa
inkeys=iaxtel
[iaxtel-outbound]
type=peer
username=swoolley
secret=gl0bal
host=iaxtel.com
The good news is that dialing 700-XXX- numbers (at Digium) works
great.
I however have two problems:
1) if I dial 800 numbers, like (800)555-1212, I get a bunch of silence
and the following in my log:
-- Starting simple switch on 'Zap/97-1'
-- Executing NoOp(Zap/97-1, ) in new stack
-- Executing Goto(Zap/97-1, intern-post|818005551212|1) in new
stack
-- Goto (intern-post,817005556226,1)
-- Executing Dial(Zap/97-1,
IAX2/myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 69.73.19.178 (format GSM)
-- Format for call is GSM
-- IAX2[iaxtel-outbound]/3 stopped sounds
The call never seems to go through.
2) Not knowing any other way to test, I have simply picked up my
asterisk SIP and analog phones and dialed my own 700 number
(700)555-6226 to which I get a bunch of silence and the following in my
log:
-- Executing NoOp(Zap/97-1, ) in new stack
-- Executing Goto(Zap/97-1, intern-post|817005556226|1) in new
stack
-- Goto (intern-post,817005556226,1)
-- Executing Dial(Zap/97-1,
IAX2/myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 69.73.19.178 (format GSM)
-- Format for call is GSM
-- IAX2[iaxtel-outbound]/2 stopped sounds
-- Hungup 'IAX2[iaxtel-outbound]/2'
But I do get a:
-- Registered to '69.73.19.178', who sees us as 63.143.35.201:4569
When asterisk is starting up so I belive I am registered.
Can I simply not dial my own 700 number from the same asterisk PBX as
a test or do I have some real problem?
--
Steve Woolley
IT Manager
ADS Telecom, Inc.
59 Skyline Drive
Suite 1250
Lake Mary, Florida 32746
Phone: (407)682-6226 x1110
Fax: (407)682-3455
Cell: (321)229-5311
[EMAIL PROTECTED]
www.adstelecom.com
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--
Mike Benoit [EMAIL PROTECTED]
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[Asterisk-Users] Broken pipe in remote exeute
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I used to be able to run asterisk -rx 'stop gracefully' on stable. But now with CVS-HEAD-07/07/04-20:09:43 it's returning: 'Broken pipe' Any ideas why, or how to fix it? - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA9AcvljK16xgETzkRAu5sAKCu/5dEwRVDjmOJeUTK61yI7czhIgCfQeXH lhMEFpU+4fzlXTlocEWd200= =OdwJ -END PGP 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] Dial Fail - Send Email
Does anyone have such an app? You might want to write this as a perl, python, php etc script using the Asterisk AGI feature. It's quite simple, after all. ___ 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] WiSIP and Zyxel Prestige 2000W
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Anyone have any experience with either of these, I 'd appreciate some feedback? Plus it seems pretty easy to steal a connection with this. Zyxel Prestige 2000W WiSIP thanks, - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA9Ah3ljK16xgETzkRApPFAJwO1PQ/5k+6UVWQaSHf6pSclg5n4wCg4SLF ZEW+HkD3RwKvEuZp42lLUBA= =2xOg -END PGP 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] Indications missing on Cisco FXO - * (SIP)
Fran Boon wrote: Calling from a Cisco FXO port to an ATA-186 (SIP 3.1 image) via * (either CVS-HEAD-06/28/04-11:43:41 or CVS-HEAD-07/12/04-15:49:58) I didn't hear any ringing sound get the following on the console: -- Called 5503 -- SIP/5503-f6b5 is ringing WARNING[-1323201616]: channel.c:1375 ast_indicate: Unable to handle indication 3 for 'SIP/10.10.2.250-9903' -- SIP/5503-f6b5 answered SIP/10.10.2.250-9903 Looking at channel.c, I can see that this means that 'condition' is neither of 'AST_CONTROL_PROGRESS' or 'AST_CONTROL_PROCEEDING'. Presumably it's 'AST_CONTROL_RINGING', so why is this not handled? (NB Calls go through fine - all ulaw currently) Further to this, I have done more digging - it's not related to the ATA at all, but is due to the Cisco FXO port. (Calls to ATA from Firefly/IAX work fine, Calls from FXO to Firefly/IAX give this same error) I have looked at Cisco's docs they talk about using progress_ind to tune which IE is sent, but this only works for H.323, not SIP: http://cisco.com/en/US/products/sw/iosswrel/ps1839/products_command_reference_chapter09186a00800b350f.html#70 Anyone using Cisco FXO ports SIP with * getting indications? Anyone using H.323 having better luck? (If so, chan_h323 or chan_oh323?) It looks to me like a bug in * as to why this IE isn't being handled, but I could be wrong. Comments welcome :) F ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 calls through IAXTEL.com
On Tue, 2004-07-13 at 05:38, Steve Woolley wrote: 1) if I dial 800 numbers, like (800)555-1212, I get a bunch of silence and the following in my log: -- Starting simple switch on 'Zap/97-1' -- Executing NoOp(Zap/97-1, ) in new stack -- Executing Goto(Zap/97-1, intern-post|818005551212|1) in new stack -- Goto (intern-post,817005556226,1) -- Executing Dial(Zap/97-1, IAX2/myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called myusername:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 69.73.19.178 (format GSM) -- Format for call is GSM -- IAX2[iaxtel-outbound]/3 stopped sounds The call never seems to go through. I am also having that problem. I started having that problem within the past few days. Maybe there's some problem at iaxtel? 2) Not knowing any other way to test, I have simply picked up my asterisk SIP and analog phones and dialed my own 700 number (700)555-6226 to which I get a bunch of silence and the following in my log: You can try calling me at 700-650-4330 and see if you have a problem and help me see if I got my IAX2 setup correct. :) -- PhoneBoy ___ 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] chan_oh323
ldconfig, check that /etc/ld.so.conf have path to where oh323 library is and then run ldconfig De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Fathallah SoumayaEnviado el: Martes, 13 de Julio de 2004 12:27 p.m.Para: [EMAIL PROTECTED]Asunto: Re: [Asterisk-Users] chan_oh323 Hello, I have been trying for a while to make the oh323 channel working but i didnt manage, i have everything compiled correctly but asterisk find somethign like an "undefined symbol" when it loads the oh323 module... i dont know if u have seen this before, I am deseperate to find the solution , i am involved in a very important project and i am out of time :( I would be very grateful if you can help me... Best Regards, soumayaLars Degenhardt [EMAIL PROTECTED] wrote: Hello,has anybody managed to register with two gatekeepers usingchan_oh323?Lars___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 Créez gratuitement votre Yahoo! Mail avec 100 Mo de stockage ! Créez votre Yahoo! Mail Dialoguez en direct avec vos amis grâce à Yahoo! Messenger !
RE: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
I have Asterisk running against 5060 and 5061 servers: [general] Port=5060 register = 8775551212:[EMAIL PROTECTED] register = 18005551212:[EMAIL PROTECTED]:5061 And then [sip-vonage] secret=secret username=18005551212 host=sphone.vopr.vonage.net port=5061 type=peer nat=yes canreinvite=no dtmfmode=rfc2833 fromuser=18005551212 context=incoming fromdomain=sphone.vopr.vonage.net [sip-bv1] secret=secret username=8775551212 host=sip.broadvoice.com Port=5060 type=peer nat=yes canreinvite=no dtmfmode=inband fromuser=8775551212 callerid=8775551212 context=incoming fromdomain=sip.broadvoice.com Runs without a problem. It's conceivable that you can run SER on port 5061 and tell Asterisk to register/dial using localhost:5061. -Original Message- From: Sunrise Ltd [mailto:[EMAIL PROTECTED] Sent: Tuesday, July 13, 2004 2:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous If Asterisk is directed to speak SIP on port 5061 and SER remains on port 5060, then how do you get Asterisk to talk to SER and vice versa? ___ 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] chan_oh323
Hi, Hello, I have been trying for a while to make the oh323 channel working but i didnt manage, i have everything compiled correctly but asterisk find somethign like an undefined symbol when it loads the oh323 module... Put the path to your openh323 libraries in your LD_LIBRARY_PATH environmental variable. You can put export LD_LIBRARY_PATH=/path/to/openh323/libs in your asterisk startup script. Jeremy ___ 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: incoming calls on Cisco 7960
From: Randy Bush [EMAIL PROTECTED] [214] disallow=all allow=ulaw type=friend secret= host=dynamic nat=no dtmfmode=rfc2833 canreinvite=no incominglimit=1 mailbox=214 where is the context= to send it to an incoming context? In the general part I have context=from-sip I don't have separate contexts for each SIP device due to the way I have this configuration set up. ___ 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] 'Dropping voice to exceptionally long queue on IAX2'
I searched the archives and found a couple references to this from Mark. I am on a network that will not support IRC (govt) so I cannot do IRC. This happend after calling Voicemail (num). I hung up on the call and also tried the # to hang up. Both resulted in many, many 'Dropping voice to exceptionally long queue on IAX2' until I physically hung up the line. It seems the detection of the hangup on voicemail using the hard hangup (handset) even took a good 10-15 seconds to register. I am running Asterisk CVS-07/13/04-09:44:05 with my patch to app_voicemail that I put on the list earlier. I have also tried the unpatched app_voicemail and have the same results. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Kannaiyan Natesan wrote: (B (BHave you used 5 welcome service in fwd? (BIf not try that. You are invited to join as a volunteer (Bto provide support and talk to new people on fwd. (B (BAsterisk can do that much better than SER because it has (Bgot a queue management system built-in. (B (BAs I explained to you before we use it for our customer (Bservice in call (Bcenter and implemented in many call centres which really (Bmakes $. (B (BNo matter how many times you claim that parallel forking (Bis the right solution to implement a call centre, it (Bdoesn't change the fact that you are still wrong. Call (Bcentres use queue management systems. (B (BCan you help me to know how that be achieved with * (Balone. (B (BSure. Read /etc/asterisk/queues.conf and (B/etc/asterisk/agents.conf. Everything you need is there. (B (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed
I work for a CLEC in Dallas where I'm presently utilizing an Option 11 and *. If you want to send the config files, I'll be happy to take a look at them and see if I can spot any inadequacies. - Original Message - From: asterisk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 13, 2004 9:38 AM Subject: RE: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed The config files would be great, thanks ! I'll let you know how I get on :) Julian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: 13 July 2004 15:16 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed We've successfully integrated with an Option 61c, but it was painful. We've set up both ends to emulate a 5eSS switch. The Asterisk is using pri_net (meaning the Nortel is pri_cpe (Client Side)). Unfortunately, in this configuration the Nortel thinks that this trunk is connected to an external phone company, so it always sends it's external Caller ID to it. That means that when someone on the Nortel calls someone on the Asterisk, you will always see the external caller id, not the actual extension from which the call originated. Our company uses Qwest to administer the Nortel, so it was the Qwest technicians who actually installed the card, set up the trunk and established the dial plan. We also found that we had to buy a special software option call Custom Dial Plan (CDP)which cost an extra $6,795 including installation. With CDP installed on the Nortel, they were able to create a dial-plan where extensions from 4000 to 4999 were sent down the trunk that's connected to the Asterisk. Asterisk then routes them accordingly. Asterisk has a dial-plan where all extensions from 2000 to 2999 are sent back to the Nortel. In that case, caller id works as it should for both name and number. Asterisk is also configured to send toll calls through the Nortel and that works correctly. So, that's the summary of what we've been able to accomplish. I can provide you with the config files on the Asterisk, but you'll need a Nortel tech for the rest; I have neither the ability nor the access to make those types of changes to a Nortel system. I hope this helps. Joe [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: Tuesday, July 13, 2004 7:11 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Meridian Option 11c Asterisk Expert Needed I've tried to do it myself, but my head is now bleeding from hitting it on the wall so much. We need someone who knows asterisk and Meridian PRI cards to help! If required, we will pay for a day's consultancy in order to get this thing working. Or, do I need to scrap my plans to keep the meridian system (60 phones ...) ... Please say no .. :) Please contact me offline (asterisk at dotr dot com) if you want to sell yourselves :) Julian. ___ 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 --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.712 / Virus Database: 468 - Release Date: 6/27/2004 ___ 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 ___ 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] OH323 and G729
Yes, it's work, Thanks, But possible don't use Global Var?, due in this situation all other destinations use this codec, after 1 time global setup. And g729 - limited:( Regards, Serge. - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 13, 2004 1:46 PM Subject: Re: [Asterisk-Users] OH323 and G729 Try with 'SetGlobalVar' instead of 'SetVar'. Michael. Serge wrote: Dear All, I have problem with new oh323 0.6.3a , I try use var OH323_OUTCODEC, but it don't work. oh323 driver don't want connect to gateway with g729, it's work if I only use in oh323.conf one codec ( g729 ). If I enable 2 or more codecs - always in use other codec: -- Executing SetVar([EMAIL PROTECTED]/1, OH323_OUTCODEC=g729a) in new stack -- Executing Dial([EMAIL PROTECTED]/1, OH323/##|70) in new stack -- H.323 call to # with codec GSM Due Gateway don't support GSM and ulaw, always return: No one is available to answer at this time Many thanks for your help, Regards, Serge. ___ 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 and Swissvoice
Does anyone have any experience with using a Swissvoice SIP phone with asterisk. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Girish Gopinath wrote:
(B
(B[globals]
(BSERADDRESS=XXX.XXX.XXX.XXX:5060
(B
(B[context]
(Bexten =
(Byourexten,1,Dial(SIP/[EMAIL PROTECTED],20,r)
(B
(BIn ser.cfg:
(B
(Bif (method == "INVITE") {
(Bif (uri =~ "sip:[EMAIL PROTECTED]"){
(Blog(1, "Forwarding to Asterisk?n");
(Brewritehostportt("XXX.XXX.XXX.XXX:5061");
(Bt_relay();
(Bbreak;
(B}
(B}
(B
(BOK, that looks kind of promising.
(B
(BIn other words you're saying to do away with
(Bauthentication for calls between Asterisk and SER since
(Bboth run on the same box?
(B
(BI haven't looked at it this way, but I guess it makes
(Bsense.
(B
(B
(BUnfortunately though, this doesn't seem to also be a
(Bsolution for what I would like to do, which is run X-Lite
(Band Asterisk on the same box, my Powerbook G4, so I can
(Buse it as an IAX phone when travelling.
(B
(Bthanks anyway for the interesting insight.
(B
(Brgds
(Bbenjk
(B
(B__
(BDo You Yahoo!?
(Bhttp://bb.yahoo.co.jp/
(B
(B___
(BAsterisk-Users mailing list
(B[EMAIL PROTECTED]
(Bhttp://lists.digium.com/mailman/listinfo/asterisk-users
(BTo UNSUBSCRIBE or update options visit:
(B http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: saving/restoring sipura config
Sorry for this OT but I bet someone here knows if there is a way to save a Sipura 2000 current config and restoring it after a reset. Thanks in advance ___ 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] WARNING: Deprecated incominglimit and outgoinglimit
For those that don't read every line of source code here's something I found out today... Deprecated incominglimit and outgoinglimit Incominglimit = number of calls the local extension can originate to Asterisk. Outgoinglimit = number of calls Asterisk will terminate to local extension. End of Life for these commands announced**, please use setgroup and checkgroup, that will also be helpful with cross channels. There is an example on how to do this at Asterisk cmd SetGroup. It's from the viewpoint of the Asterisk PBX, not from the local extension. The CLI command sip show inuse will show the current status. The outgoinglimit is currently disabled in the source code of the SIP channel. -- ** someone is kidding here, right? Announced? I think not... Andy ___ 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] Installing Digiums Dev Kit Lite
Im really new to asterisk and linux and im having a problem getting this all working. I installed asterisk and copied all the conf files I got with the devkitlite disk.. Ive installed the PCI card and the USB s100u, I then ran modprobe wcfxo and modprobe wcusb, but when I run ztcfg I get this message: ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that fxs interfaces re configured with fxo signalling and that fxo interfaces use fxs signalling? Im running redhat, and im lost. Any help would be greatly appreciated from anyone who has played with this kit before.. Thanks -chad ___ 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: Help Needed in configuring Cisco 7940
What I had to do was to download and install the 5.0 image, provision the phone and then upgrade it to the latest release. Kurt __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ 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] G729A and GSM - newbie question
On Tue, 2004-07-13 at 10:44, Oleg A. Arkhangelsky wrote: Hello, When I'm trying to play standard sound files from Asterisk using G729A codec with OH323 channel I get this message: channel.c:1650 ast_set_write_format: Unable to find a path from GSM to G729A It seems that this files must be in G729 format? How can I convert this files to G729? ... or am I wrong? http://www.google.com/search?hl=enq=g729+site%3Alists.digium.com Use google please. From that link you should quickly figure out that you must purchase a license for G729. G729 is a codec not a file format so you still wouldn't get anywhere with the question you asked. -- Steven Critchfield [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] Re: Cann't load oh323 0.6.3a
Hi Oleg, Yes, you are right. I havn't executed ldconfig. Thank you very much. But now after I added the path /usr/local/lib to /etc/ld.so.conf and started asterisk, there is another error: [chan_oh323.so]Jul 13 12:56:47 WARNING[1074464512]: loader.c:240 ast_load_resource: /usr/local/lib/liboh323wrap.so: undefined symbol: _ZTI14PAbstractArray Jul 13 12:56:47 WARNING[1074464512]: loader.c:421 load_modules: Loading module chan_oh323.so failed! After I executed ldconfig, I didn't edit the driver's configuration file. Should I firstly edit the configration file? Could you give me another help? Thanks a lot. Rui Oleg A. Arkhangelsky wrote: Hello ruixun, Tuesday, July 13, 2004, 6:26:53 PM, you wrote: rw [chan_oh323.so]Jul 13 09:43:45 WARNING[1074460416]: rw loader.c:240 rw ast_load_resource: liboh323wrap.so: cannot open shared rw object file: No rw such file or directory Have you executed ldconfig after installing oh323? By default liboh323wrap.so is located in /usr/local/lib and you must add this path to /etc/ld.so.conf. __ Post your free ad now! http://personals.yahoo.ca ___ 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] fax still fails, ideas sought! Re: rxfax/spandsp fails to decode
Okay having taken in some suggestions and googled this topic to death I'm still stuck - anyone got any ideas? To recap, the faxes are coming in via a digium E1 card but failing to train properly or if they manage it sending a garbled and very truncated fax. A number of folks have suggested clock sync issues.. my zaptel.conf is set to use the PRI as primary clock, i have no evidence of issues altho dont know how to check (other than the call quality is fine, no clicks, no pri down/ups). What can i try? Steve On Mon, 12 Jul 2004, Stephen J. Wilcox wrote: Hi, I just sent this to Steve Underwood, but then found a bunch of posts on the mailing list about similar issues.. does anyone have the fix? I'm running asterisk CVS-HEAD-06/28/04-18:13:13, spandsp 0.0.1k, libtif 3.5.7 one thing i just noticed is that calls come in with format '72' which is G711A-law or LinearPCM.. it uses PCM for the call, i assume this is ok the results of RxFAX vary, it sometimes saves the file in which case i get errors: Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 0 (got 2383, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 1 (x 137). and the resulting tif looks to be only a few rows long or more commonly it just fails entirely.. i paste the output below so you can see. is there anything obvious i'm doign wrong here? TIA! Steve. -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/faxes/20040712-183339.tif) in new stack Changed from phase 0 to 1 Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 31 37 31 31 36 35 34 35 34 38 30 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: DCS: 83 00 86 90 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 5ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1699.90 (64) Training error 56.874846 Training succeeded (constellation mismatch 44.212022) Fast carrier trained Fast carrier down Trainability test failed - longest run of zeros was 14 FTT: 44 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.33 (64) Training error 51.989152 Training succeeded (constellation mismatch 37.988826) Fast carrier trained Fast carrier down Trainability test failed - longest run of zeros was 15 FTT: 44 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.32 (64) Training error 60.898646 Training succeeded (constellation mismatch 46.138793) Fast carrier trained Fast carrier down Trainability test failed - longest run of zeros was 17 FTT: 44 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1795.61 (4) Fast carrier down Fast carrier up Coarse carrier frequency 1789.60 (4) Fast carrier down -- Channel 0/1, span 1 got hangup -- Hungup 'Zap/1-1' ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous
Olle E. Johansson wrote: (B (BWell, I have users that get an account on my PBX. (B (BI really don't care how many phones they want to use, (Bhardware phones on their desktop or soft phones on their (Blaptop while travelling. It's still a user with one (Baccount. (B (BTwo words: self provisioning. (B (BAsterisk doesn't really bother with *users*, it has a (Bdevice-centric view of life, universe and propably (Beverything. (B (BThat's only partly correct. The queue management system (Bhas a user view, called agents, and agents can (Bauthenticate themselves independently from the device they (Bare using and then attach themselves to call queues (Bmanaged by Asterisk. (B (BHowever, for anything unrelated to queue management, you (Bare correct in that Asterisk doesn't apply this concept (Bthere. (B (BI may even agree with you that it would be worthwhile to (Bapply this user concept to other areas outside of queue (Bmanagement. (B (BStill I disagree that parallel forking is the way to do (Bthis. I even disagree that it would introduce a user view. (BInstead it would water down the device view. So you go (Bfrom an system with a very clean device view but without a (Buniversally applied user view to a system with a messy (Bdevice view and still no user view. (B (BWith Asterisk, the user has to call me each time he wants (Ba new (Bdevice connected and I have to reconfigure his setup. (B (BNot if you give them a means to provision it themselves. (BThis can be as easy as an extension that asks for a PIN (Bnumber and then executes a shell script. (B (BIf I had support for multiple registrations on one [peer] (Baccount, the (B[peer] would become a user account instead of a device (B (BWell, that's an opinion. (B (BI'd rather prefer to have a user layer on top and in (Baddition to a device layer instead of trading one for the (Bother. (B (BThis is how GSM works BTW, you have the IMEI which (Bidentifies the device and the IMSI which identifies the (Bsubscriber. A subscriber may be using the same IMSI on (Bdifferent devices, but the IMEI for each device is unique. (BThe IMEI lives in the device. The IMSI lives on the SIM (Bcard. (B (BThe customer care and billing system is mostly concerned (Babout the IMSI when dealing with a subscriber, but some (Blow level network elements need the IMEI to do their job. (BThe conclusion here is that there is a use for both, (Bdevice and user views. (B (BI think it would be wise to take a lesson from GSM in (Brespect of having both a device and a user view, and not (Bjust trade one for the other. (B (BAnd the user could add as many devices as he wanted (B(up to a defined limit) without bothering the (Badministrator. (B (BEarly mobile phone systems made the same mistake you are (Bproposing here. They too said "device = user" and it (Bopened the door to plenty of problems, from inconvenience (Bwhen changing a device to fraud. (B (BThe introduction of GSM introduced a user layer on top of (Bthe device layer and you got both convenience (ie move the (BSIM card to another phone and secondary SIM for a family (Bmember etc etc) and better security (no device cloning, (Bstolen equipment can be blocked through EIR network (Belements etc etc). (B (B (BI guess that's why a lot of people ask for this function. (B (BNo, people asking for this because "If all you have is a (Bhammer, everything looks like a nail." (B (BHowever, since Asterisk doesn't really bother with a user (Bconcept, (Bwe really have to teach Asterisk about users. And user (Bgroups. (B (BI agree with that in priniciple, but parallel forking (Bdoesn't do that. (B (BI've been discussing this many times, and so has many (Bother (Bpeople. I think we need an elegant way of defining users (Bto (Basterisk so we connect peers, users, agents and mailboxes (Bto a *user* with one set of credentials. If you look into (Byour (BAsterisk configuration, you will find that there are (Busers and (Bcredentials for logging in everywhere. It's not easy to (Bmaintain at all. (B (BAgreed again, but still fail to see how parallel forking (Bwould contribute anything to what you ask for here. (B (BHint: I have a new idea for a solution on multiple reg's. (BRaise the bounty and I might give it a try. ;-) (B (BIf you absolutely have to mess with it, just make sure it (Bcan be disabled by the rest of us who don't want to deal (Bwith any potential problems it may introduce. (B (Brgds (Bbenjk (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP: One way audio... continuously and randomly
On Tue, 2004-07-13 at 11:50, Vasyl Rublyov wrote: :( Just getting silence Is this mailing list alive at all? Suggestion: o Move your existing src to an archive folder. o Download new cvs code and compile. o Using new default config files, start with default codecs and very simple configs. o Slowly add back your settings and test between each change till you find where the problem is. :) Maybe there is a simple problem somewhere that you will find. Vasyl Rublyov wrote: All, I seen already threads about one way audio... but never seen anyone answered completely on it. There is a problem, what we are getting, even with stable-1, CVS updates in May, June as well as last Saturday (Jul 10, 2004) [T1/PRI PSTN] == [Lucent Legend PBX] == [T1/PRI] == [T100P Asterisk IAX2] == [T1 Internet (ISP Verizon = QWest) connected thru T100P interfaces (before it was NetOpia T1 router but the same problem existed)] === [ADSL Internet (ISP: UTEL/Ukraine)] === [IAX2: Asterisk with TDM400 cards] === [Analog phones SIP phones (Cisco 79xx Polycom IP500] -- respectfully, Joseph - (606) 477-2355 x140 --= ___ 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
