[Asterisk-Users] make sipura stop generating stale nonce. Device comes in and goes out every 1 minute
I am authenticating sipura device as a sip user to my asterisk server. Things work fine and then suddenly asterisk console tells me: Oct 26 23:09:17 WARNING[5096]: chan_sip.c:4826 check_auth: Stale nonce received from 'Sipura1PSTN sip:[EMAIL PROTECTED]' as soon as that happens if i try to call this sipura device extension number it tells me: -- Executing Dial(IAX2/[EMAIL PROTECTED]/5, SIP/994|20) in new stack -- Called 994 -- Got SIP response 410 Gone back from 59.xx.xxx.xxx -- SIP/994-cb94 is circuit-busy and the call does not work. The call starts to work again when on the asterisk console i see: -- Registered SIP '994' at 59.xx.xxx.xx port 1249 expires 30 I have tried changing the following values in the sipura admin interface: register retry interval - 1 seconds. register retry long term interval - 30 seconds But still it is generating the stale nonce. How to make the sipura device stop generating the stale nonce, Please help, s ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] faxdetect on voicemail
hi, is there anyway to just enable faxdetection in voicemail? thanks, paradise dove ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Incoming calls via CAPI and AVM Fritz Card
Hi,
I'm not really sure if this helps you, but as far as I remember, the
diastring with chan_capi-cm-0.6 is not
CAPI/g1/0299546476:b${EXTEN},30,r but
CAPI/ggroup/destination[/params] or in your case
CAPI/g1/${EXTEN}/b,30,r.
To set your CallerPresentation, use the SetCallerPres() in your
Dialplan, which is now used as the CLIP/CLIR.
Regards Jörg
Esteban Guana-Jarrin wrote:
Can anyone please provide some help. I have installed an AVM fritz card
on an asterisk box ([EMAIL PROTECTED] version 1.5). I have installed the
card driver and chan_capi-cm-0.6. According to the installations guide I
can now see that the CAPI channel in asterisk is up,
*CLI capi info
Contr1: 2 B channels total, 2 B channels free.
I set up a trunk and the dialstring includes the following,
CAPI/g1/0299546476:b${EXTEN},30,r
My capi.conf is,
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
;ulaw=yes;set this, if you live in u-law world instead of a-law
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;ntmode=yes ;if isdn card operates in nt mode, set this to yes
isdnmode=msn ;'MSN' (point-to-multipoint) or 'DID' (direct inward dial)
;when using NT-mode, ptp should be set in any case
incomingmsn=*;allow incoming calls to this list of MSNs/DIDs, * == any
;controller=0;ISDN4BSD default
;controller=7;ISDN4BSD USB default
controller=1 ;capi controller number to use
group=1 ;dialout group
;prefix=0;set a prefix to calling number on incoming calls
softdtmf=on ;enable/disable software dtmf detection, recommended
for AVM cards
relaxdtmf=on ;in addition to softdtmf, you can use relaxed dtmf
detection
accountcode= ;Asterisk accountcode to use in CDRs
context=from-trunk
;context=capi-in ;context for incoming calls
holdtype=hold;when Asterisk puts the call on hold, ISDN HOLD will be
used. If
;set to 'local' (default value), no hold is done and
Asterisk may
;play MOH.
;immediate=yes ;immediate start of pbx with extension 's' if no digits
were
;received on incoming call (no destination number yet)
;echosquelch=1 ;_VERY_PRIMITIVE_ echo suppression
;echocancel=yes ;EICON DIVA SERVER (CAPI) echo cancelation
;(possible values: 'no', 'yes', 'force', 'g164', 'g165')
echocancelold=yes;use facility selector 6 instead of correct 8
(necessary for older eicon drivers)
;echotail=64 ;echo cancel tail setting
;bridge=yes ;native bridging (CAPI line interconnect) if available
;callgroup=1 ;Asterisk call group
;deflect=1234567 ;deflect incoming calls to 1234567 if all B channels
are busy
devices=2;number of concurrent calls on this controller
;(2 makes sense for single BRI, 30 for PRI)
I can't see that a number is assigned to msn, but I read somewhere on
this list that for this latest version of chan_capi this is not required.
I connected the asterisk box to the ISDN line, which belongs to a Hunt
group with number as shown in the dialstring and when ringing that
number from an external line I do not get any tone and asterisk does not
log any indications of incoming calls via the CAPI channel
Can anyone please shed some light on what do I need to do in order to be
able to receive calls via this setup.
Thanks in advance,
PolAUs
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[Asterisk-Users] please recommend phones with adsi.
Hello, can somebody recommend me any hard or may be even softphones which support ADSI. I would like to work with Asterisk voicemail application using ADSI. Thanks, Dmitry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 + No Hangup
Any problems with bristuff ? 2005/10/26, Julian J. M. [EMAIL PROTECTED]: You can try this patch (www.maxosystem.net/asterisk/asterisk-stable-polarity.html), if your telco sends your polarity reversals on answer and hangup. Julian J. M. On 10/26/05, Giovanni Miano [EMAIL PROTECTED] wrote: I've TDM400P with 2 cards fxo and asterisk 1.0.9 + zaptel 1.0.9.2 All works perfectly but command Hangup or Hangup() in dialplan dont hangup call (zapata.conf within busycount=4 and busydetect=yes) Why ? Country is ITALY -- Giovanni Miano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sKinny in database
Hi, Isit possible to make the skinny working over a odbc/mysql/oracle db? what i have to put in the extconfig.conf and how must the tables look like? Hope somebody can help me.. thx rene ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 + No Hangup
No... It applies without problems (just a little offset) Julian. On 10/27/05, Giovanni Miano [EMAIL PROTECTED] wrote: Any problems with bristuff ? 2005/10/26, Julian J. M. [EMAIL PROTECTED]: You can try this patch (www.maxosystem.net/asterisk/asterisk-stable-polarity.html), if your telco sends your polarity reversals on answer and hangup. Julian J. M. On 10/26/05, Giovanni Miano [EMAIL PROTECTED] wrote: I've TDM400P with 2 cards fxo and asterisk 1.0.9 + zaptel 1.0.9.2 All works perfectly but command Hangup or Hangup() in dialplan dont hangup call (zapata.conf within busycount=4 and busydetect=yes) Why ? Country is ITALY -- Giovanni Miano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Trying to genereate dial tone, but stop after first digit dialed.
I don't know if this will relate to your specific issue, but I had problems with system not responding to numbers I pressed right away when dialing internally-i.e. the dialtone did not stop like it should when system reads numbers pressed (DTMF). I found that adjusting the rxgain and txgain in the etc/asterisk/zapata.conf file affected the responses. The defaults for both are 0.0 and apparantly represent percentage. Too high makes feedback, too low and system will not respond. With my TDM11B I have rxgain at 6.0 and txgain at 3.0 right now, but I am still working out playback volume and IVR responses. that's all I got Neil T. Skowronek --- Jonathan Feally [EMAIL PROTECTED] wrote: I seem to be missing something here. Basically I'm trying to do what a full CO would do in terms of *70 to disable call waiting. I have a *70 exten setup, it does the work to set the extension to not take in a second call, then does a playtones(dialrecall). This works except that all digits dialed after the *70 have the tone still playing until the dialplan kicks back in for the new exten dialed. Does somebody have a work around for this? I'd prefer to not use Background. Thanks, -Jon ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Need to ztcfg every time I reboot *
Hi Angus, I have the same problem but on a Debian distro I do not know very well... When I boot the machine only wcfxs and zaptel modules are loadedhow can I load qozap before wcfxs? TIA Giorgio Angus Comber wrote: Hello I am sure this is a very basic Linux question. But every time I reboot my * I need to modprobe module and then ztcfg After doing this I can then run * without it complaining about not loading a channel. The module being loaded is qozap - a ISDN card. What do I need to do to make the ztcfg configuration persistent? Angus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Need to ztcfg every time I reboot *
On Thu, Oct 27, 2005 at 10:42:18AM +0200, gincantalupo wrote: Hi Angus, I have the same problem but on a Debian distro I do not know very well... When I boot the machine only wcfxs and zaptel modules are loadedhow can I load qozap before wcfxs? echo qozap /etc/modules I figure that wcfxs is modprobed by hotplug and that modprobe loads zaptel with it. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 iptables rules
One last check...won't ask again, promise :) Does someone know a solution to my problem below? Best Regards Goran - Original Message - From: Goran Tornqvist To: asterisk-users@lists.digium.com Sent: Wednesday, October 26, 2005 10:33 AM Subject: Asterisk iptables rules Hello, I have trouble getting asterisk to work with my new firewall script (see below). I used this info as base: 'http://www.voip-info.org/wiki-Asterisk+firewall+rules And then modified it to suit my needs. I use only SIP and the problem is that the calls get in to asterisk when the firewall is activated. But my agents/phones cant register or receive any calls. So all callsget stuck in queue on asterisk. So I believe Im missing some rule perhaps? Can anyone help me sortthis out? Thanks... Best Regards Goran /etc/init.d/firewall == #IPTables firewall configuration for X export PATH=$PATH:/sbin case "$1" in start) echo "Starting iptables firewall..." iptables --flush iptables --delete-chain iptables -A INPUT -p icmp -i eth0 -j ACCEPT # START OPEN PORTS #= #SSH (22) iptables -A INPUT -p tcp -i eth0 --dport 22 -j ACCEPT #SAMBA: netbios (139) , microsoft-ds (445) iptables -A INPUT -p tcp -i eth0 --dport 139 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 445 -j ACCEPT #ASTERISK # SIP (UDP 5060) iptables -A INPUT -p tcp -m tcp -i eth0 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5060 -j ACCEPT # IAX2/IAX iptables -A INPUT -p udp -m udp -i eth0 --dport 4569 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5036 -j ACCEPT # RTP - the media stream iptables -A INPUT -p udp -m udp -i eth0 --dport 1:2 -j ACCEPT # MGCP - if you use media gateway control protocol in your configuration iptables -A INPUT -p udp -m udp -i eth0 --dport 2727 -j ACCEPT #END ASTERISK #MySQL (3306) iptables -A INPUT -p tcp -i eth0 --dport 3306 -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 3306 -j ACCEPT #SNMP (161) - Allow from cacti server iptables -A INPUT -p tcp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT #Ftp / Passive ports iptables -A INPUT -p tcp -i eth0 --dport 21 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 64785:64799 -j ACCEPT #Http / Web iptables -A INPUT -p tcp -i eth0 --dport 80 -j ACCEPT #Webmin (1) iptables -A INPUT -p tcp -i eth0 --dport 1 -j ACCEPT # END OPEN PORTS #= #Deny everything else iptables -A INPUT -p all -i eth0 -j DROP exit 0; ;; stop) echo "Stopping iptables firewall..." iptables --flush iptables --delete-chain exit 0; ;; *) echo "Valid switches: firewall start , firewall stop"; esac; ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 iptables rules
I would suggest that you are missing something like: iptables -A INPUT -m state --state ESTABLISHED,RELATED -i eth0 -j ACCEPT This will mean that if a UDP packet is sent by * from sport:2345, dport:5060, then the response (sport:5060, dport:2345) will be allowed in, whereas at present that is not the case. I cannot say whether this type of packet will ever be sent, but I always include the rule for completeness. Alternatively, add a LOG rule, just before the DROP rule, and see what is being dropped... Regards, Steve On 10/27/05, Goran Tornqvist [EMAIL PROTECTED] wrote: One last check...won't ask again, promise :) Does someone know a solution to my problem below? Best Regards Goran - Original Message - From: Goran Tornqvist To: asterisk-users@lists.digium.com Sent: Wednesday, October 26, 2005 10:33 AM Subject: Asterisk iptables rules Hello, I have trouble getting asterisk to work with my new firewall script (see below). I used this info as base: 'http://www.voip-info.org/wiki-Asterisk+firewall+rules And then modified it to suit my needs. I use only SIP and the problem is that the calls get in to asterisk when the firewall is activated. But my agents/phones cant register or receive any calls. So all calls get stuck in queue on asterisk. So I believe Im missing some rule perhaps? Can anyone help me sort this out? Thanks... Best Regards Goran /etc/init.d/firewall == #IPTables firewall configuration for X export PATH=$PATH:/sbin case $1 in start) echo Starting iptables firewall... iptables --flush iptables --delete-chain iptables -A INPUT -p icmp -i eth0 -j ACCEPT # START OPEN PORTS #= #SSH (22) iptables -A INPUT -p tcp -i eth0 --dport 22 -j ACCEPT #SAMBA: netbios (139) , microsoft-ds (445) iptables -A INPUT -p tcp -i eth0 --dport 139 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 445 -j ACCEPT #ASTERISK # SIP (UDP 5060) iptables -A INPUT -p tcp -m tcp -i eth0 --dport 5060 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5060 -j ACCEPT # IAX2/IAX iptables -A INPUT -p udp -m udp -i eth0 --dport 4569 -j ACCEPT iptables -A INPUT -p udp -m udp -i eth0 --dport 5036 -j ACCEPT # RTP - the media stream iptables -A INPUT -p udp -m udp -i eth0 --dport 1:2 -j ACCEPT # MGCP - if you use media gateway control protocol in your configuration iptables -A INPUT -p udp -m udp -i eth0 --dport 2727 -j ACCEPT #END ASTERISK #MySQL (3306) iptables -A INPUT -p tcp -i eth0 --dport 3306 -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 3306 -j ACCEPT #SNMP (161) - Allow from cacti server iptables -A INPUT -p tcp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT iptables -A INPUT -p udp -i eth0 --dport 161 --source x.x.x.x -j ACCEPT #Ftp / Passive ports iptables -A INPUT -p tcp -i eth0 --dport 21 -j ACCEPT iptables -A INPUT -p tcp -i eth0 --dport 64785:64799 -j ACCEPT #Http / Web iptables -A INPUT -p tcp -i eth0 --dport 80 -j ACCEPT #Webmin (1) iptables -A INPUT -p tcp -i eth0 --dport 1 -j ACCEPT # END OPEN PORTS #= #Deny everything else iptables -A INPUT -p all -i eth0 -j DROP exit 0; ;; stop) echo Stopping iptables firewall... iptables --flush iptables --delete-chain exit 0; ;; *) echo Valid switches: firewall start , firewall stop; esac; ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] open-source vs. tellme/skype/gNumber et. al.
Now that Skype and Ebay are one, I feel that they will be cherry-picking all the promising open-source voip/asterisk development and calling it their own. There is a company called gNumber that relies completely on Asterisk that has also teamed up with ebay for cell phone notification of ending auctions, they claim to have patents pending on 'transactions through voice channel' I'm new to open-source, perhaps this is the wrong forum to ask this question but where does the line exist between shared and ownership? The software that is asterisk has allowed for all this to develope, can people then take what freely distrubuted and own it? I know I'm opening a can of worms and need to read more on this, but I'd like to hear some learned opinions first and at least get a few links to help me with my research. THX Neil __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] open-source vs. tellme/skype/gNumber et. al.
On Thu, October 27, 2005 12:10, Neil Skowronek said: Now that Skype and Ebay are one, I feel that they will be cherry-picking all the promising open-source voip/asterisk development and calling it their own. There is a company called gNumber that relies completely on Asterisk that has also teamed up with ebay for cell phone notification of ending auctions, they claim to have patents pending on 'transactions through voice channel' I'm new to open-source, perhaps this is the wrong forum to ask this question but where does the line exist between shared and ownership? The software that is asterisk has allowed for all this to develope, can people then take what freely distrubuted and own it? I know I'm opening a can of worms and need to read more on this, but I'd like to hear some learned opinions first and at least get a few links to help me with my research. THX Neil You sure this isn't a homework question? :-) Very short answer: The GPL only allows use and redistribution of any of the source released under the GPL if it remains under the GPL. So *legally* they cannot take GPL code and call it their own. Whether or not it happens is hard to say, esp. with closed source... But that is only the tip of the iceberg. I hope it helps you along to start the proper research... Google is your friend, as is http://www.gnu.org/copyleft/gpl.html -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Overlap dial and match as you go = how to implement early dial on telco line
Hi, I have Asterisk between PBX and telco line. PBX is reporting number in overlap dial manner. I'd like to early connect to telco line as soon as I get for instance two numbers, that distinguish telco calls. But the problem is if I receive 3 numbers at once, then two numbers dialplan rule will not be matched I've found some references to similar problems, but I'm not sure which solution was included in Asterisk (if any) So I'd kindly ask if anyone has working solution or has idea how to do this on recent Asterisk to describe it... Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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+Nat+sipura (Help)
Hi ALL; I have users with Sipura/Linksysphones regsitered behind Nat( useing STUNat phonenot portforwarding) in my Asterisk box, when I try to call them with another phone i got: Got SIP response 404 "Not Found" back from 217.6.190.4 SIP/217.6.190.4:5060-666d is circuit-busy Isabove mentioned problem relates to "Nat", Is there anybody who use sipura with STUN method and can recive calls? My asterisk Sip.conf for Nat has the following: [sipura] .. nat=yes canreinvite=no qualify=1000 Appreciate any help Mohammad ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SPA3000 as trunk - no caller ID
Kerry Garrison wrote: During a PSTN call the status screen correctly displays the caller ID information. Well, if the SPA-3000 is picking up the CID, and PSTN CID as VOIP CID is set, and the caller ID isn't being passed to Asterisk, it looks as if the SIP INVITE is being passed to Asterisk before the CID has been detected. But you've obviously thought of that - hence the delay... It may be worth firing up ethereal to check that the CID really isn't in the INVITE. Are you using version 3.1.7 of the Sipura firmware? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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+Nat+sipura (Help)
I don't think the problem is NAT-related. Looks like To header in SIP INVITE message do not match to User ID in sipura settings. On Thu, 2005-10-27 at 01:57 +0330, mohammad mirzaee wrote: Hi ALL; I have users with Sipura/Linksysphones regsitered behind Nat( useing STUNat phonenot portforwarding) in my Asterisk box, when I try to call them with another phone i got: Got SIP response 404 Not Found back from 217.6.190.4 SIP/217.6.190.4:5060-666d is circuit-busy Isabove mentioned problem relates to Nat, Is there anybody who use sipura with STUN method and can recive calls? My asterisk Sip.conf for Nat has the following: [sipura] .. nat=yes canreinvite=no qualify=1000 Appreciate any help Mohammad ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Echo - Solved with KB1 Patch
I've had an absoloutely fantastic run with the new KB1 patch currently on mantis - http://bugs.digium.com/view.php?id=5520 The Digium guys are looking for feedback, please apply and test - If we can get some positive feedback, it might make it into 1.2! --Rob ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems compiling asterisk zaptel for Asterisk 1.0.9
Hello all. I have installed the Asterisk 1.0.9. But I am facing problems compiling the zaptel for asterisk. I am getting lots of errors stating dereferencing pointer to incomplete type. The error appears in the zaptel.c file. Could anybody please let me know if they have come across the same error? And also could anybody suggest me any solution for the same. Thanks in advance Regards, Bharat M. Sarvan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Polycom 601 XHTML microbrowser
Chris HARIGA wrote: Gary Reuter wrote: On 10/26/05, *Chris HARIGA* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have a show parked calls php script for my Polycom IP600 phones. If U are interested let know and I can email it. Even if Sean doesn't want it, I do! All examples can be helpful. :-) Why not put up a page on the wiki linked from the polycom page(s)... If formatting is problematic, just note it on the page and I (and others) can help make look nicer for the wiki. -Gary ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I will edit the wiki and I will upload my polycom scripts: parked calls, sip users status, meetme status, queues list and phones status tonight. Best regards, Chris HARIGA Please! I've bee wondering if anything was available along these lines. All that space on the LCD with nothing to do! This will be of huge benefit to a large number of people - thanks you. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
Hi list, i'm having a problem with asterisk+pstn termination, i just bought a TDM400p and connect my phone line(bellsouth) now when im using the pstn through asterisk there's an echo, i don't know if this is already have been resolved. If it does please point me to the instruction how to resolve this. Try reading README.fxotune and using fxotune to see if it improves it. If you do an fxotune and all of the coefficients are 0, does this mean that fxotune is not making any changes? I've got 6 lines that are coming from a channel bank into two TDM cards and have significant echo, even with Asterisk HEAD and KB1. I just ran fxotune, and all 6 lines came back with all 0's in fxotune.conf... Based on what Matt has mentioned previously, fxotune only sets the impedence to proper values today. He has not implemented the code to set the coefficients as yet, therefore the expected values are zero's. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp / txfax exit codes / logging?
Is it possible to somehow read spandsp / txfax exit codes? What I mean, I never know if the fax sent through the Asterisk box was sent successfully, or not (i.e., a real person picked up the phone instead of a fax machine). A possibility of reading an exit code, or a log file would allow to build some kind of fax confirming (via email/web page/etc.). Are exit codes (or logging, or something similar) possible with spandsp / txfax? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] spandsp / txfax exit codes / logging?
I'm looking for that one too. I had not been succesfull up to now. Bob. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tomasz Chmielewski Sent: Thursday, October 27, 2005 1:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] spandsp / txfax exit codes / logging? Is it possible to somehow read spandsp / txfax exit codes? What I mean, I never know if the fax sent through the Asterisk box was sent successfully, or not (i.e., a real person picked up the phone instead of a fax machine). A possibility of reading an exit code, or a log file would allow to build some kind of fax confirming (via email/web page/etc.). Are exit codes (or logging, or something similar) possible with spandsp / txfax? -- Tomek http://wpkg.org WPKG - software deployment and upgrades with Samba ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] tellme/skype voice apps go live
On Wed, 26 Oct 2005, Dean Collins wrote: Thought this may be of interest to some people on this list. https://studio.tellme.com/skype/submissionprocess.html Bullet point 4 translates for me into If you live in South Africa or another country where Paypal won't take customers, go away now and don't bother us. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Message Waiting Indicator and PRI
Hi I have a pri connection working on asterisk; I would like to send the MWI on the PRI link Libpri code clearly says that it is there, but there is no document in asterisk says anything about this. The current mailbox config also doesnt work Anyone has any idea about this? Cheers Mustafa N. Deeb ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Simple SIP only Asterisk Configuration
That shouldnt be complicate, but it looks like you re not registering with your provider. However, without the configuration files, it is not much to do for help you. Carlos Alperin From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pikoro Sent: Wednesday, October 26, 2005 11:01 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Simple SIP only Asterisk Configuration Ok, I've been messing with asterisk for the last 3 weeks and I just can't seem to figure this out. Everything I've read seems to state that asterisk will work out of the box with only minor config changes when being used only for SIP to SIP calls. The problem I am having is I cannot make outbound calls or receive incoming calls over my sip-provider. Asterisk registers properly, and internal communications seem to work fine. I have, at one time or another, had either outgoing only, or incoming only, but never both at once. Unfortunately, I didn't know what I did to make either of those work since I had made multiple adjustments and had done a reload after each change. For some reason, the incoming calls only started working after restarting the computer so it could have been any of 50 things I had changed. I am back to the sample config files. Is there any kind of walkthrough for a sip only setup? I have seen SIP only touched on briefly, with most of the documentation leaning torwards IAX communications. Here is what I am trying to accomplish: Asterisk server registers with our sip-provider for sip to pstn local and international calls Internal extensions 0, 200-210 can call eachother (of course) Extensions 200-205 are in a Tech Support Queue Extensions 206-210 are in a Customer Support Queue Extension 0 is the operator or menu system (I guess this would be s?) All phones (for now) are x-ten soft phones Each extension has voice mail When a customer calls during office hours, they are presented with a menu, press 1 for CS, press 2 for TS, or dial the extension you wish to reach, etc... Calls can be forwarded to other extensions On-hold music is implemented I can handle doing everything on the list except for #1. If anyone can offer any suggestions, it would make me, and my boss, very happy. Thanks in Advance Aaron ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Polycom 601 XHTML microbrowser
Faris Raouf wrote: Chris HARIGA wrote: Gary Reuter wrote: On 10/26/05, *Chris HARIGA* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I have a show parked calls php script for my Polycom IP600 phones. If U are interested let know and I can email it. Even if Sean doesn't want it, I do! All examples can be helpful. :-) Why not put up a page on the wiki linked from the polycom page(s)... If formatting is problematic, just note it on the page and I (and others) can help make look nicer for the wiki. -Gary ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi, I will edit the wiki and I will upload my polycom scripts: parked calls, sip users status, meetme status, queues list and phones status tonight. Best regards, Chris HARIGA Please! I've bee wondering if anything was available along these lines. All that space on the LCD with nothing to do! This will be of huge benefit to a large number of people - thanks you. Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://www.voip-info.org/wiki/view/Polycom+XML+browser+scripts+for+asterisk Best regards, Chris HARIGA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] smp
On Wed, Oct 26, 2005 at 03:48:19PM -0500, John HIll wrote: I have a small test system -- 6 phones. It is a dual processor server. I noticed that asterisk spawns 12 child processes. Can this be controlled? I would think 2-4 would be plenty for this test site. Asterisk generally spans a separate thread for each channel and has a number of other threads. On Linux 2.4 (without NPTL) you will see each thread as a separate process in the output of ps and top. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vmail.cgi and realtime?
I've been given the charge of finding out if anyone has gotten vmail.cgi to work with asterisk realtime, pulling the voicemail users from the db... I thank you all for any input you may have Sherwood ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problems compiling asterisk zaptel for Asterisk 1.0.9
On Thu, Oct 27, 2005 at 04:43:21PM +0530, Bharat M. Sarvan wrote: Hello all. I have installed the Asterisk 1.0.9. But I am facing problems compiling the zaptel for asterisk. I am getting lots of errors stating dereferencing pointer to incomplete type. I have a number of such warnings and they seem harmless. Does the compilation fail? If so, could you please produce a log of the build? try: make 21 | tee logfile -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: delays with IAX2 and Meetme
Hi Tony
Thanks for the reply and for
posting the code. I added the code and recompiled Asterisk, but unfortunately
it did not resolve the issue. It basically trapped all of the incoming audio
and wrote to the error log instead of outputting it. So basically it never
seemed to go to the careful_write statement.
To answer your questions:
Firstly, I am using kernel 2.6, but am not using Ztdummy. I am using a digium
card for timing. I have run a test on it, and it seems to be working properly.
I am using a client built
using IaxClient, and am now looking at the possibility that the delay might be
a client issue instead of a server issue. What is the best tool to use to run
tests on my server and clients to narrow down the source of the delay?
Many thanks
Steven
Date: Wed, 12 Oct 2005
10:41:33 + (UTC)
From:
[EMAIL PROTECTED] (Tony Mountifield)
Subject:
[Asterisk-Users] Re: delays with IAX2 and Meetme
To: asterisk-users@lists.digium.com
Message-ID:
[EMAIL PROTECTED]
In article
[EMAIL PROTECTED],
Steven Langley [EMAIL PROTECTED] wrote:
I am using IAX2
softphones dialing into meetme conferences. I also have
jitterbuffer=yes,
with typical jitterbuffer settings. The problem I am
having is that as
soon as there is a delay from a participant, then the
delay continues
until the participant hangs up and dials in again. When
dialing in again
the delay seems to go.
It seems to me as
though as soon as the server registers a delay from a
participant, then
it causes delay on all further packets from that
participant.
Does anyone have
any ideas what the problem could be?
Yes, there are a few
possibilities. Firstly, are you using ztdummy for
timing? Which kernel
version? If 2.6, have you ensured that USE_RTC is
correctly defined in
ztdummy.c?
Look in bugs.digium.com
at bug IDs 3599 and 4252 - they might be relevant.
Yesterday I found
another mechanism which could give rise to both a delay
and broken audio - I
found it with OH323 channels, but it might possibly
arise on other channel
types too. It concerns a backlog building up in
the channel driver and
never being drained by meetme because of blocking
in the pseudo-device
when trying to write the contents of a large frame.
In app_meetme.c, try
replacing this:
careful_write(fd,
f-data, f-datalen);
with this:
if (f-datalen =
CONF_SIZE) {
careful_write(fd, f-data, f-datalen);
} else {
ast_log(LOG_WARNING,
Discarding large frame (%d bytes) from channel %s\n, f-datalen,
chan-name);
}
and see if it helps.
I haven't yet submitted
the above change to mantis.
Cheers
Tony
--
Tony Mountifield
Work:
[EMAIL PROTECTED] - http://www.softins.co.uk
Play:
[EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Bristuff question
Hi there, I have 2 ISDN modems (HFC chipset). I use bristuff from junghanns. Its possible to load one cards as NT (T-Bus) and the other as S-Bus. When I do make load the 2 cards loads as S-Bus and when I do make loadNT the 2 cards loads as T-Bus. Can someone help me?? Thanks in advance Pedro Nunes ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 1.2beta and te411p: incorrectly reporting sometimes all channels busy
Hi, we have strange problem on our new card. Sometimes it reports all channels busy, so call cannot be made (it doesn't even appear in log). We've contacted Digium support, but received no usable answer (they've told us that this card should work on stable Asterisk version - AFAIK this is not correct)... Any advice, what to check and what are possible cause of such behaviour ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Bristuff question
http://www.voip-info.org/wiki-Asterisk+zaphfc look this Giordano Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Per conto di Pedro Nunes Inviato: giovedì 27 ottobre 2005 16.23 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Bristuff question Hi there, I have 2 ISDN modems (HFC chipset). I use bristuff from junghanns. Its possible to load one cards as NT (T-Bus) and the other as S-Bus. When I do make load the 2 cards loads as S-Bus and when I do make loadNT the 2 cards loads as T-Bus. Can someone help me?? Thanks in advance Pedro Nunes ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SPA3000 as trunk - no caller ID
Here is my version Software Version: 3.1.5(GWb) Hardware Version: 2.0.1(42a8) I had mentioned this before, I am downloading 3.1.7 right now. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Daragon Sent: Thursday, October 27, 2005 3:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID Kerry Garrison wrote: During a PSTN call the status screen correctly displays the caller ID information. Well, if the SPA-3000 is picking up the CID, and PSTN CID as VOIP CID is set, and the caller ID isn't being passed to Asterisk, it looks as if the SIP INVITE is being passed to Asterisk before the CID has been detected. But you've obviously thought of that - hence the delay... It may be worth firing up ethereal to check that the CID really isn't in the INVITE. Are you using version 3.1.7 of the Sipura firmware? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ANNOUNCEMENT : A2Billing - AreskiCC V3 newrelease
Senad, We welcome competition of any kind.It just makes us improve and aim higher. Ans: Before aiming higher, why dont you guysjust deliver a working software for your current clients, who paid the money and never got anything in return.? Good luck to A2Billing in its pursue although our billing module is just not the same. Ans: Correct, your billing module just does not work. Is that what you mean? In regards to your other posts in the past you will hear from our legal team soon. We are just too busy currently implementing SWITCHware, but rest be assured we will catch up with you.Ans: My Advise to you and SteveWingfield is not to imitate the Nigerian scam and make threats of harm, both online and offline to your clients, from foreign soil. We Americansknew how to deal with these threats. My advise to you is to desist from any such unproductive adventure and focus on giving back something useful to the Asterisk community, from whom you have taken so so so much thus far. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
[EMAIL PROTECTED] wrote on 10/27/2005 08:22:04 AM: If you do an fxotune and all of the coefficients are 0, does this mean that fxotune is not making any changes? Based on what Matt has mentioned previously, fxotune only sets the impedence to proper values today. He has not implemented the code to set the coefficients as yet, therefore the expected values are zero's. Are these settings persistent across reboots? The README for fxotune seems to mention that you need to do a fxotune -s in order to reload the card with the analyzed settings (rather than take the 20 minutes it seems to take on my 6 lines). However, if fxotune.conf is all 0's, I sure hope that the settings are persistent on the board! :) Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] please recommend phones with adsi.
Look on the wiki which is located at: http://www.voip-info.org/ On 10/27/05, Dmytro Mishchenko [EMAIL PROTECTED] wrote: Hello, can somebody recommend me any hard or may be even softphones which support ADSI. I would like to work with Asterisk voicemail application using ADSI. Thanks, Dmitry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Polycom 601 XHTML microbrowser
At 08:38 AM 10/27/2005, you wrote: http://www.voip-info.org/wiki/view/Polycom+XML+browser+scripts+for+asterisk Best regards, Chris HARIGA Thanks. Is it possible for someone to provide a basic explanation of how to implement this for us less technical minded people? From what I can tell, it looks like one needs to modify the ipmid.cfg file. I'm guessing the mp.proxy, mp.main.home, and/or mb.limits.nodes values need to be modified. My guess is that I simply copy the files to an appropriate folder and modify the mp.main.home setting to point to that folder. The mp.proxy and mp.limits.nodes values can be left null? Thanks, Doug ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SPA3000 as trunk - no caller ID
Upgraded to 3.1.7
Excerpts from Asterisk Log:
Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT INTO
cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
,billsec,disposition,amaflags,accountcode) VALUES ('2005-10-27
07:43:50','\Garrison Kerry\
9496799285','9496799285','s','from-sip-external',
'SIP/192.168.5.200-083279d0','','Congestion','',0,0,'NO ANSWER',3,'')
Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf(SIP/spa3000-8d99,
0?from-pstn-reghours|s|1:) in new stack
Oct 27 07:43:56 DEBUG[1531]: Check for res for spa3000
Oct 27 07:43:56 DEBUG[1531]: Call from user 'spa3000' is 1 out of 0
Oct 27 07:43:56 DEBUG[1531]: build_route: Contact hop:
Oct 27 07:43:56 DEBUG[1531]: Expression is '0'
Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf(SIP/spa3000-8d99,
0?from-pstn-reghours|s|1:) in new stack
The log is interesting in that it actually is pushing the CID across but
then something strange is happening, if I look at my CDR it shows the
following:
The call comes in to SIP/192.168.5.200 Source is the correct source phone
number, Clid is correct CID, Dst is s, Disposition is NO ANSWER
6-7 seconds later it there is another entry
The call comes in to SIP/spa3000 Source is now empty, Clid is spa3000, Dst
is 201, Disposition is ANSWERED
Here is a link to a screenshot of the SPA3000 settings:
http://techdatapros.com/temp/spa3000.gif
-Kerry
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Re: [Asterisk-Users] Vmail.cgi and realtime?
Are you having a problem? Have you even tried to do it? We are using asterisk realtime with MySQL voicemail integration. vmail.cgi works just fine. I think I had to tweak a variable in it to tell it to look in the database instead of a file. Open the CGI up and take a look at it. On 10/27/05, Sherwood McGowan [EMAIL PROTECTED] wrote: I've been given the charge of finding out if anyone has gotten vmail.cgi to work with asterisk realtime, pulling the voicemail users from the db... I thank you all for any input you may have Sherwood ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question on callingpres and blocked numbers
Kevin Bockman wrote: This is just a feature of PRI service. Of course all of the call info is still available even if you 'block' it. The call still has to be traceable. Magic huh? I thought that was cool too the first time I found out about it. It depends on whether you are purchasing retail or wholesale service (at least it is supposed to), retail meaning 'end user' and wholesale meaning 'carrier'. The presumption is that an end user can't be trusted to suppress the information just because the flag is set requesting them to do so, so their upstream provider actually removes the information but leaves the 'restricted' flag turned on so that the recipient knows why the information is not there. A 'carrier' customer is expected to provide the same service for their end-user customers, and pass along the data unmodified to other carrier customers. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Bristuff question
Giordano, Thanks, stupid question. Ive look to that page 100 of times but I do not remember that part of the page about loading more than one card :S. Thanks again From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giordano Grandis Sent: quinta-feira, 27 de Outubro de 2005 15:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: R: [Asterisk-Users] Bristuff question http://www.voip-info.org/wiki-Asterisk+zaphfc look this Giordano Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] Per conto di Pedro Nunes Inviato: giovedì 27 ottobre 2005 16.23 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Bristuff question Hi there, I have 2 ISDN modems (HFC chipset). I use bristuff from junghanns. Its possible to load one cards as NT (T-Bus) and the other as S-Bus. When I do make load the 2 cards loads as S-Bus and when I do make loadNT the 2 cards loads as T-Bus. Can someone help me?? Thanks in advance Pedro Nunes ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Test after Hurricane Wilma
Hi guys. Please disregard this. I'm testing connectivity after being down due to Hurricane Wilma. Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Register to Asterisk using MAC address.
Try Voice Over Ethernet. Asterisk cannot do that since it only supports Voice Over IP. On 10/25/05, Maps [EMAIL PROTECTED] wrote: Dear Supporters! Does any one know how to set the asterisk to allow the phone to register to asterisk using the MAC address? Thanks! Lan Phan. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
I had to turn on the aggressive echo cancellation in the zaptel drivers for mine. Which is much better, but we still get occasional pops. The funny part is only the asterisk side of the connection hears the echo. Jared Armstrong -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Thursday, October 27, 2005 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] asterisk using tdm400p has echo Hi list, i'm having a problem with asterisk+pstn termination, i just bought a TDM400p and connect my phone line(bellsouth) now when im using the pstn through asterisk there's an echo, i don't know if this is already have been resolved. If it does please point me to the instruction how to resolve this. Try reading README.fxotune and using fxotune to see if it improves it. If you do an fxotune and all of the coefficients are 0, does this mean that fxotune is not making any changes? I've got 6 lines that are coming from a channel bank into two TDM cards and have significant echo, even with Asterisk HEAD and KB1. I just ran fxotune, and all 6 lines came back with all 0's in fxotune.conf... Based on what Matt has mentioned previously, fxotune only sets the impedence to proper values today. He has not implemented the code to set the coefficients as yet, therefore the expected values are zero's. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: delays with IAX2 and Meetme
Steven Langley wrote: Hi Tony Thanks for the reply and for posting the code. I added the code and recompiled Asterisk, but unfortunately it did not resolve the issue. It basically trapped all of the incoming audio and wrote to the error log instead of outputting it. So basically it never seemed to go to the careful_write statement. To answer your questions: Firstly, I am using kernel 2.6, but am not using Ztdummy. I am using a digium card for timing. I have run a test on it, and it seems to be working properly. I am using a client built using IaxClient, and am now looking at the possibility that the delay might be a client issue instead of a server issue. What is the best tool to use to run tests on my server and clients to narrow down the source of the delay? I'm pretty sure this is a known issue with enabling meetme enter/exit sounds. Turn them off (q option to MeetMe, I think) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ANNOUNCEMENT : A2Billing - AreskiCC V3 newrelease
Seshu, So, now you are not Seshu Kanuri any more but Pbxware Swithware? Since you are not working or associated with our company I need to ask you not to use Pbxware, Switchware in your email client From field. PBXware SWITCHware wrote: Senad, We welcome competition of any kind. It just makes us improve and aim higher. Ans: Before aiming higher, why dont you guys just deliver a working software for your current clients, who paid the money and never got anything in return.? That maybe in your case since you destroyed your copy. In regards to your other posts in the past you will hear from our legal team soon. We are just too busy currently implementing SWITCHware, but rest be assured we will catch up with you. Ans: My Advise to you and Steve Wingfield is not to imitate the Nigerian scam and make threats of harm, both online and offline to your clients, from foreign soil. We Americans knew how to deal with these threats. We are not making threats... You WILL hear from our lawyers. My advise to you is to desist from any such unproductive adventure and focus on giving back something useful to the Asterisk community, from whom you have taken so so so much thus far. Yes you are right. Our solutions add so much more to GPL software. That is what is all about... BUILDING for better future, our kids etc. Is it not? We will not go back to stone age and re-invent the wheel. Obviously you do! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SPA3000 as trunk - no caller ID
Kerry Garrison wrote: Upgraded to 3.1.7 Excerpts from Asterisk Log: Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT INTO snip ... Here is a link to a screenshot of the SPA3000 settings: http://techdatapros.com/temp/spa3000.gif I get connection refused at that URL. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voip asterisk second edition
I have the first edition, does anyone know if it's worth getting the second too? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] smp
Tzafrir, Thanks for the reply. This is a 2.6.13 kernel. Runs very well. It really is not hurting anything memory usage is ok and it is responsive. Just my old school resource attitude. Shana Tova --john -- This mail was scanned by AntiVir Milter. This product is licensed for non-commercial use. See www.antivir.de for details. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: delays with IAX2 and Meetme
StevenThere are issues being looked at, see: http://bugs.digium.com/view.php?id=3599http://bugs.digium.com/view.php?id=4252 Always worth while checking through bugs.digium.comRegardsRobOn 10/27/05, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:Steven Langley wrote: Hi Tony Thanks for the reply and for posting the code. I added the code and recompiled Asterisk, but unfortunately it did not resolve the issue. It basically trapped all of the incoming audio and wrote to the error log instead of outputting it. So basically it never seemed to go to the careful_write statement. To answer your questions: Firstly, I am using kernel 2.6, but am not using Ztdummy. I am using a digium card for timing. I have run a test on it, and it seems to be working properly. I am using a client built using IaxClient, and am now looking at the possibility that the delay might be a client issue instead of a server issue. What is the best tool to use to run tests on my server and clients to narrow down the source of the delay?I'm pretty sure this is a known issue with enabling meetme enter/exitsounds.Turn them off (q option to MeetMe, I think)___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SPA3000 as trunk - no caller ID
I've had a very similar thing on my SPA-3000 and they only way to fix
it was a full default reset on the SPA and reconfigure it from scratch
8-(
Matt.
On 27/10/05, Kerry Garrison [EMAIL PROTECTED] wrote:
Upgraded to 3.1.7
Excerpts from Asterisk Log:
Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT INTO
cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration
,billsec,disposition,amaflags,accountcode) VALUES ('2005-10-27
07:43:50','\Garrison Kerry\
9496799285','9496799285','s','from-sip-external',
'SIP/192.168.5.200-083279d0','','Congestion','',0,0,'NO ANSWER',3,'')
Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf(SIP/spa3000-8d99,
0?from-pstn-reghours|s|1:) in new stack
Oct 27 07:43:56 DEBUG[1531]: Check for res for spa3000
Oct 27 07:43:56 DEBUG[1531]: Call from user 'spa3000' is 1 out of 0
Oct 27 07:43:56 DEBUG[1531]: build_route: Contact hop:
Oct 27 07:43:56 DEBUG[1531]: Expression is '0'
Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf(SIP/spa3000-8d99,
0?from-pstn-reghours|s|1:) in new stack
The log is interesting in that it actually is pushing the CID across but
then something strange is happening, if I look at my CDR it shows the
following:
The call comes in to SIP/192.168.5.200 Source is the correct source phone
number, Clid is correct CID, Dst is s, Disposition is NO ANSWER
6-7 seconds later it there is another entry
The call comes in to SIP/spa3000 Source is now empty, Clid is spa3000, Dst
is 201, Disposition is ANSWERED
Here is a link to a screenshot of the SPA3000 settings:
http://techdatapros.com/temp/spa3000.gif
-Kerry
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Re: [Asterisk-Users] please recommend phones with adsi.
I sure like my Aastra 390, the voicemail ADSI app works pretty well (only a couple of incompleted functions, like not exiting by hanging up the speakerphone, rather than go to a reorder tone. As for the 'look at the wiki' comment, I'm not trying to get on anyone's badside, but Dmitry was asking for recommendations, not documentation. Sorry, not pointing fingers, but I see that 'blanket answer' of going to to Wiki all too often on here lately ;) If I was asking for recommedations of good jazz music, it wouldnt help me to have someone tell me to go to Sam Goody. Chris - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, October 27, 2005 7:53 AM Subject: Re: [Asterisk-Users] please recommend phones with adsi. Look on the wiki which is located at: http://www.voip-info.org/ On 10/27/05, Dmytro Mishchenko [EMAIL PROTECTED] wrote: Hello, can somebody recommend me any hard or may be even softphones which support ADSI. I would like to work with Asterisk voicemail application using ADSI. Thanks, Dmitry ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ANNOUNCEMENT : A2Billing - AreskiCC V3 newrelease
Take this silly argument off-line please. On Sun, 27 Nov 2005 16:28:27 -, Senad Jordanovic wrote: Seshu, So, now you are not "Seshu Kanuri" any more but "Pbxware Swithware"? Since you are not working or associated with our company I need to ask you not to use "Pbxware, Switchware" in your email client "From" field. PBXware SWITCHware wrote: Senad, We welcome competition of any kind. It just makes us improve and aim higher. Ans: Before aiming higher, why dont you guys just deliver a working software for your current clients, who paid the money and never got anything in return.? That maybe in your case since you destroyed your copy. In regards to your "other" posts in the past you will hear from our legal team soon. We are just too busy currently implementing SWITCHware, but rest be assured we will catch up with you. Ans: My Advise to you and Steve Wingfield is not to imitate the Nigerian scam and make threats of harm, both online and offline to your clients, from foreign soil. We Americans knew how to deal with these threats. We are not making threats... You WILL hear from our lawyers. My advise to you is to desist from any such unproductive adventure and focus on giving back something useful to the Asterisk community, from whom you have taken so so so much thus far. Yes you are right. Our solutions add so much more to GPL software. That is what is all about... BUILDING for better future, our kids etc. Is it not? We will not go back to stone age and re-invent the wheel. Obviously you do! ___ --Bandwidth and Colocation sponsored by Easynews.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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SPA3000 as trunk - no caller ID
I just tested it from a different location without any problem. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Daragon Sent: Thursday, October 27, 2005 8:44 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID Kerry Garrison wrote: Upgraded to 3.1.7 Excerpts from Asterisk Log: Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT INTO snip ... Here is a link to a screenshot of the SPA3000 settings: http://techdatapros.com/temp/spa3000.gif I get connection refused at that URL. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Have IAXy signal busy without losing ongoing call?
I'm using an IAXy witha current CVS-head build of Asterisk. The IAXy has an extensions.conf entry somethng like this: exten = 1,1,Ringing exten = 1,2,Answer exten = 1,3,Voicemail(u1) exten = 1,4 Hangup This works fine for calls routed to extension 1. But if a second call is routed to the IAXy while it's already busy, the first call disappears from the IAXy and the second one goes to voicemail. Asterisk says that the second call is accepted by the IAXy and then it determines that the extension is busy. But when the second call is accepted, the first silently disappears. Is doesn't seem to get hung-up until I hang-up the phone attached to the IAXy. Is there a way to have Asterisk dtermine that the IAXy is busy without interrupting the ongoing IAXy call? frank ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Polycom 601 XHTML microbrowser
[EMAIL PROTECTED] wrote: At 08:38 AM 10/27/2005, you wrote: http://www.voip-info.org/wiki/view/Polycom+XML+browser+scripts+for+asterisk Best regards, Chris HARIGA Thanks. Is it possible for someone to provide a basic explanation of how to implement this for us less technical minded people? From what I can tell, it looks like one needs to modify the ipmid.cfg file. I'm guessing the mp.proxy, mp.main.home, and/or mb.limits.nodes values need to be modified. My guess is that I simply copy the files to an appropriate folder and modify the mp.main.home setting to point to that folder. The mp.proxy and mp.limits.nodes values can be left null? Thanks, Doug Um, well the easiest thing to do is: 1) stick the files on your webserver somewhere (e.g. www.mydomain.com/pcom) 2) Modify the top lines of each .php file so that the ip address is that of your asterisk server, and the username and password match a username and password configured in manager.conf 3) Change the config on your polycom phone via the web browser rather than hacing away at the xml. Once logged in, click on the microbrowser link option (I think it is in the general section), leave the proxy server line blank, and just put www.mydomain.com/pcom or wherever in as the address. Click on OK. The phone reboots and the xml config files will automatically update (assuming you allow TFTP uploads on your TFTP server). And then it just works! Faris. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ANNOUNCEMENT : A2Billing - AreskiCC V3 newrelease
George Gardiner wrote: Take this silly argument off-line please. Yap.. you are right.. it should not be here... apologies! Senad ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SPA3000 as trunk - no caller ID
RELLLY???
Hell, I can do that. Anything is worth a try at this point. I have it fully
documented so restoring the settings shouldn't take but a few minutes. I am
just not going to be in the office for about 5 hours now and not going to
ask my wife to do it. I will certainly try it, its had half a dozen firmware
updates and a bajillion setting changes, it certainly wont hurt to try it.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of InetUID
Sent: Thursday, October 27, 2005 9:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID
I've had a very similar thing on my SPA-3000 and they only way to fix it was
a full default reset on the SPA and reconfigure it from scratch 8-(
Matt.
On 27/10/05, Kerry Garrison [EMAIL PROTECTED] wrote:
Upgraded to 3.1.7
Excerpts from Asterisk Log:
Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,du
ration
,billsec,disposition,amaflags,accountcode) VALUES ('2005-10-27
07:43:50','\Garrison Kerry\
9496799285','9496799285','s','from-sip-external',
'SIP/192.168.5.200-083279d0','','Congestion','',0,0,'NO ANSWER',3,'')
Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf(SIP/spa3000-8d99,
0?from-pstn-reghours|s|1:) in new stack Oct 27 07:43:56 DEBUG[1531]:
Check for res for spa3000 Oct 27 07:43:56 DEBUG[1531]: Call from user
'spa3000' is 1 out of 0 Oct 27 07:43:56 DEBUG[1531]: build_route:
Contact hop:
Oct 27 07:43:56 DEBUG[1531]: Expression is '0'
Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf(SIP/spa3000-8d99,
0?from-pstn-reghours|s|1:) in new stack
The log is interesting in that it actually is pushing the CID across
but then something strange is happening, if I look at my CDR it shows
the
following:
The call comes in to SIP/192.168.5.200 Source is the correct source
phone number, Clid is correct CID, Dst is s, Disposition is NO ANSWER
6-7 seconds later it there is another entry The call comes in to
SIP/spa3000 Source is now empty, Clid is spa3000, Dst is 201,
Disposition is ANSWERED
Here is a link to a screenshot of the SPA3000 settings:
http://techdatapros.com/temp/spa3000.gif
-Kerry
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Re: [Asterisk-Users] Zaptel stop hangs server
[EMAIL PROTECTED] wrote on 10/26/2005 06:53:51 PM: I have two TE110P cards. If I stop the Zaptel service, the whole server hangs. I have had this issue with 1.0.7, 1.0.8 ,1.0.9 and 1.0.9.2. The server is a Dell 1750 with all unnecessary BIOS options off (USB, Serial, Second NIC, etc) It is Dual CPU. There are no shared Interrupts. please advise if anyone has had this issue and figured it out. I had this same problem... I found that using zaptel 1.2.0 beta1 works fine. I contacted Digium support and detailed the problem using the TE110p and zaptel 1.0.x. I also let them know that modprobe'ing and rmmod'ing using zaptel 1.2.0 beta1 worked fine. They told me it was safe to use the 1.2.0 beta1 version with libpri 1.0.9 and asterisk 1.0.9 using the TE110p. I am a little nervous about running this way, however, it does seem to work. I offered to let someone log into my machine to debug this issue and was told that since the machine locks it would be to hard to debug this problem and just go ahead and use the 1.2.0 beta1 version. Seems like someone would have been interested as to why this config doesn't work with STABLE zaptel. Anyway, hope this helps. Bill ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: delays with IAX2 and Meetme
Hi Thanks for the reply I do actually use the |q option to disable the enter/exit sounds. Steven Message: 15 Date: Thu, 27 Oct 2005 10:25:32 -0500 From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: delays with IAX2 and Meetme To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Steven Langley wrote: Hi Tony Thanks for the reply and for posting the code. I added the code and recompiled Asterisk, but unfortunately it did not resolve the issue. It basically trapped all of the incoming audio and wrote to the error log instead of outputting it. So basically it never seemed to go to the careful_write statement. To answer your questions: Firstly, I am using kernel 2.6, but am not using Ztdummy. I am using a digium card for timing. I have run a test on it, and it seems to be working properly. I am using a client built using IaxClient, and am now looking at the possibility that the delay might be a client issue instead of a server issue. What is the best tool to use to run tests on my server and clients to narrow down the source of the delay? I'm pretty sure this is a known issue with enabling meetme enter/exit sounds. Turn them off (q option to MeetMe, I think) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Network Architecture Question
I currently have the following network configuration: Internet--Firewall --- DMZ --- Company A --- Company B --- Company C Each company has its own network address I want to install asterisk and use SIP hardware phones that will be located in all the companies and also have a few phones that will be located on remote sites (only 2 or 3). I have been thinking that the best place to put the server will be in DMZ. Am I right or you can suggest me some other solution? Thanks. Make sure YOUR emails don't get lost! Download Mailinfo here ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] x100p (FXO) not being seen by asterisk (is my bestguess) .
Hello Phil , On Thu, 27 Oct 2005, Phil Pritchard wrote: only new to asterisk, but have had some hardware exp. stay away from irq9 its tied to irq2 and will always be shared, Paul has the go.. in bios disable serial and or usb (if not using) and make sure irda is not enabled. another one is the lpt port if your not using that, there is another irq you can steel.. ALL I mean all serial/parrallel/...'everything I can find'... has been turned off in the bios . And I have recompiled a kernel with those same items turned off in it . That d??ned module wants to load at irq 9 no matter what I do . Of course there is no way to set irq's to a particular pci slot in the bios . Does anyone now howto set irq say at the boot: or in modprobe.conf ? dont share interrupts, as a rule(if you can help it)... it usually leads to system instability and usually under load. Quite well understand this point . Have heard it on this list many times . And am doing my best NOT too . UBCD ...(www.ultimatebootcd.com). has some nice tools that can probe a system to give a second appinion on interrupt conflicts, ram and hard drive errors. its my best tool for hardware problems.. IMO , The mirrors have the su??iest download schemes I have seen in some time .\IMO I have yet to burn that image but as soon as I do I'll boot it on that piece of junk I bought for near next to nothing . Which is almost what it is worth , Nothing . Thank you for your input , Every bit helps . JimL Mr. James W. Laferriere wrote: Hello Paul all , On Wed, 26 Oct 2005, Mr. James W. Laferriere wrote: Hello Paul all , I've tried everything I know to attempt to get the wcfxo.ko not to use irq 9 . THe 6 line cord does not appear to effect the signaling to the x100p card , I have turned up the debugging have that being syslog'd . Have debugging on zaptel as well . Nothing seems out of the ordinary . But monitoring from 'asterisk -d -v -nr'console does not show anything '.' . Have I forgotten some configurations or magical incantation ? Tia , JimL On Wed, 26 Oct 2005, Paul wrote: First I don't like the 6 line cord. Use an rj11 2 wire cord, but watch the crossover vrs straight on the old red and green. Next the interrupt must be fixed. Do this in the CMOS before you boot. Go to the PCI bus assignments and set the IRQ or go and disable the serial ports thereby allowing irq 3 and 4 to be assigned. :) Paul Sorry about the top posting ... Also forgot the syslog output . Tia , JimL Oct 26 20:11:19 asterisk-test kernel: kobject zaptel: registering. parent: NULL, set: module Oct 26 20:11:19 asterisk-test kernel: subsystem zaptel: registering Oct 26 20:11:19 asterisk-test kernel: kobject zaptel: registering. parent: NULL, set: class Oct 26 20:11:19 asterisk-test kernel: kobject zaptimer: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: kobject zapchannel: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: kobject zappseudo: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: kobject zapctl: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: Zapata Telephony Interface Registered on major 196 Oct 26 20:11:19 asterisk-test kernel: kobject wcfxo: registering. parent: NULL, set: module Oct 26 20:11:19 asterisk-test kernel: kobject wcfxo: registering. parent: NULL, set: drivers Oct 26 20:11:19 asterisk-test kernel: PCI: Found IRQ 9 for device :01:02.0 Oct 26 20:11:19 asterisk-test kernel: PCI: Sharing IRQ 9 with :00:1f.3 Oct 26 20:11:19 asterisk-test kernel: kobject zap1: registering. parent: zaptel, set: class_obj Oct 26 20:11:19 asterisk-test kernel: New regoffset: 7 Oct 26 20:11:20 asterisk-test kernel: wcfxo: DAA mode is 'FCC' Oct 26 20:11:20 asterisk-test kernel: Found a Wildcard FXO: Wildcard X101P Oct 26 20:11:20 asterisk-test kernel: Recalculating slaves on WCFXO/0/0 Oct 26 20:11:20 asterisk-test kernel: Done Recalculating slaves on WCFXO/0/0 (last is WCFXO/0/0) Oct 26 20:11:20 asterisk-test kernel: Configured channel WCFXO/0/0, flags 0201, sig 2004 Oct 26 20:11:20 asterisk-test kernel: Registered tone zone 0 (United States / North America) Oct 26 20:11:20 asterisk-test kernel: BATTERY! Oct 26 20:11:39 asterisk-test kernel: Out of storage space Oct 26 20:11:48 asterisk-test kernel: RING! Oct 26 20:11:50 asterisk-test kernel: NO RING! Oct 26 20:11:54 asterisk-test kernel: RING! Oct 26 20:11:56 asterisk-test kernel: NO RING! Oct 26 20:13:50 asterisk-test kernel: RING! Oct 26 20:13:52 asterisk-test kernel: NO RING! Oct 26 20:13:56
[Asterisk-Users] Re: Zaptel stop hangs server
I'll give it a shot. Do you compile it with Zaptel running or diasable it and reboot first? -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] [EMAIL PROTECTED] wrote on 10/26/2005 06:53:51 PM: I have two TE110P cards. If I stop the Zaptel service, the whole server hangs. I have had this issue with 1.0.7, 1.0.8 ,1.0.9 and 1.0.9.2. The server is a Dell 1750 with all unnecessary BIOS options off (USB, Serial, Second NIC, etc) It is Dual CPU. There are no shared Interrupts. please advise if anyone has had this issue and figured it out. I had this same problem... I found that using zaptel 1.2.0 beta1 works fine. I contacted Digium support and detailed the problem using the TE110p and zaptel 1.0.x. I also let them know that modprobe'ing and rmmod'ing using zaptel 1.2.0 beta1 worked fine. They told me it was safe to use the 1.2.0 beta1 version with libpri 1.0.9 and asterisk 1.0.9 using the TE110p. I am a little nervous about running this way, however, it does seem to work. I offered to let someone log into my machine to debug this issue and was told that since the machine locks it would be to hard to debug this problem and just go ahead and use the 1.2.0 beta1 version. Seems like someone would have been interested as to why this config doesn't work with STABLE zaptel. Anyway, hope this helps. Bill ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: delays with IAX2 and Meetme
In article [EMAIL PROTECTED], Steven Langley [EMAIL PROTECTED] wrote: Hi Tony Thanks for the reply and for posting the code. I added the code and recompiled Asterisk, but unfortunately it did not resolve the issue. It basically trapped all of the incoming audio and wrote to the error log instead of outputting it. So basically it never seemed to go to the careful_write statement. That's strange. The log message tells you the size of the incoming audio frame. What size frames are you getting? To answer your questions: Firstly, I am using kernel 2.6, but am not using Ztdummy. I am using a digium card for timing. I have run a test on it, and it seems to be working properly. OK, hardware timing should be fine. I am using a client built using IaxClient, and am now looking at the possibility that the delay might be a client issue instead of a server issue. What is the best tool to use to run tests on my server and clients to narrow down the source of the delay? The first thing would be, can you set the audio frame size in your IAX client? What codec are you using? If you can, try ulaw or alaw with 20ms frames. Not sure about the best tool to track this down Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cannot get dialtone or ring on FXS ports (TDM400p)
Just added a TDM400p with 2 fxs ports to asterisk so that I could hook up our fax lines, ztcfg shows the card being detected and configured correctly (fxo_ks signalling) Zapata.conf [channels]context=incomingsignalling=pri_cpeswitchtype=nationalechocancel=yesechocancelwhenbridged=yesechotraining=400callerid=asreceivedgroup=1channel = 1-12 context=analogintsignalling=fxo_kslanguage=engroup=2channel = 15-16 also "zap showchannel 15" shows no problems, likewise with "zap show status" slots 34 on the card are populated...dialing the channels produces no rings, and no errors no dialtone. the card used to have fxo modules (I switched them) aux power connector is attached. I'm stuck, any help would be appreciated. Thanks, Jay Bhatt ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] QoS Monitor
I would like to be able to monitor my QoS.. I see that Qwest is using this QoS Manager (Firehunter) http://www.home.agilent.com/cgi-bin/pub/agilent/Product/cp_Product.jsp?NAV_ID=-536885714.536882909.00LANGUAGE_CODE=engCONTENT_KEY=49888ID=49888COUNTRY_CODE=US I have some buddies who work at Qwest and use this software, however they are monitoring primarly Sonus GSX switches with it, has anyone used this in an asterisk environment? -=Linsys=- IntrusionSec.com #1 Hacker Gamez Web Site On the Internet http://www.intrusionsec.com [EMAIL PROTECTED] - When Your Life Flashes Before Your Eyes When You Die, Does That Include The Part Where Your Life Flashes Before Your Eyes? - ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Echo Canceller question- is there a viable solution?
On Oct 27, 2005, at 12:38 AM, [EMAIL PROTECTED] wrote: My question is, what is the direction in relation to analog boards and such? Right now, it looks like the current fad of the asterisk group is hardware echo cancelation. However, there is work that is occurring on the software echo cans to improve them. In fact, I just committed basically an update to KB1 (which was until now the latest and greatest version of MEC2) that is supposed to provide somewhat significant improvements. Quite a few people tend to have difficulties with echo, and although the WIKI has some very helpful advice, from a business standpoint I would think that it would be an important step to come up with a final solution to the problem. Many companies who make the higher end equipment seem to have tackled the issue on their hardware. Do we know if digium is spending time on solving the issue? For example, having a tool to run on a digium analog or t1 board to analyze the line statistics and come up with the proper gain settings could be extremely helpful. Such a tool would require a firm knowledge of the causes and solutions to echo however, but I would assume that digium should have a grasp on this. It just seems difficult to suggest to companies to use an asterisk based solution (if they do not use pri) when there is the possibility that an installation will have issues with echo. At this point, it feels more like a trial experience to eliminate echo in various environments. Unfortunately, that's the way it is right now. Getting to the point where you have enough knowledge to be able to work on these things is not an insignificant task. It seems like we're slowly getting there, and now that we have some more interest on improving the software echo cans we might be a little be closer to getting to the point where it just works. I have used local tone from the CO to help narrow things down, but a tool that would loop dial a line and do an analysis could reduce the implementation time from days to hours. Well, there isn't anything that does the whole job right now. There's a bunch of pieces that go together, and if you have the necessary knowledge of how to put the pieces together, you can get pretty close to it just working. It's not that bad though, one can also see it as job security as well :-) Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Message Waiting Indicator and PRI
On Oct 27, 2005, at 8:04 AM, Mustafa N. Deeb wrote: I have a pri connection working on asterisk; I would like to send the MWI on the PRI link Libpri code clearly says that it is there, but there is no document in asterisk says anything about this. The current mailbox config also doesn’t work Anyone has any idea about this? Libpri supports it, however it is not implemented in Asterisk. It only works on Q.SIG signaled lines as well. Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
On Oct 27, 2005, at 12:18 AM, [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] wrote on 10/26/2005 05:09:30 PM: On Oct 26, 2005, at 3:40 PM, Mark Quitoriano wrote: Hi list, i'm having a problem with asterisk+pstn termination, i just bought a TDM400p and connect my phone line(bellsouth) now when im using the pstn through asterisk there's an echo, i don't know if this is already have been resolved. If it does please point me to the instruction how to resolve this. Try reading README.fxotune and using fxotune to see if it improves it. If you do an fxotune and all of the coefficients are 0, does this mean that fxotune is not making any changes? I've got 6 lines that are coming from a channel bank into two TDM cards and have significant echo, even with Asterisk HEAD and KB1. I just ran fxotune, and all 6 lines came back with all 0's in fxotune.conf... Try the new echo canceller in head. It's an update to KB1 called MG2. You'll have to enable it in zconfig.h kind like with KB1. You can post feedback to bug #5120 on mantis. Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM01B vs. X100P
Hi, I apologize in advance if this is a stupid question, but I have not been able to find an answer by searching the web. I would like to add an FXO port or two to my Asterisk setup, and I am wondering if there is any good reason to spend $120 on a TDM01B or $180 on a TDM02B instead of paying $9.95 or $19.90 for one or two new, genuine, unopened X100P cards on eBay. I am not particularly worried about running out of PCI slots, as I don't envision ever needing to add any other line cards to this machine. However, if there is some kind of substantial quality compatibility difference between the two cards, I would like to know about this before wasting (even a small amount of) money on X100Ps. Thanks, Rusty ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
On Oct 27, 2005, at 12:02 AM, Mark Quitoriano wrote: i tried doing the instruction from voip-info[1] anyway here's my comment with that instruction. when i tried doing /usr/src/zaptel/fxotune -i 4 it gives me this Tuning module 1Failure! Tuning module 2Failure! Tuning module 3Failure! Tuning module 4Failure! how can i debug this? i look at my /var/log/messages and it gives me many of this line Oct 26 17:36:06 sloan kernel: — Set echo registers successfully Oct 26 17:36:25 sloan kernel: — Setting echo registers: I'm using AAH 1.5 stock no modifications. I think they're using versions 1.0.9 for all(e.g. asterisk, zaptel). Don't use 1.0.9. It's old. Try the version of fxotune from head. Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fxotune fails with valid TDM/FXO card
Fxotune doesn't appear to work with the latest TDM boards. I have a TDM400P rev I card and receive the following when running fxotune : # ./fxotune -i 4 Tuning module 1 Skipping non-TDM / non-FXO Failure! Tuning module 2 Skipping non-TDM / non-FXO Failure! I didn't see anything obvious in the code that ties this to the card revision, but I recalled seeing something on the list about previous changes in this regard. Any suggestions on getting this to work? System details : Fedora Core 4 Kernel 2.6.13-1.1526_FC4smp Asterisk CVS-v1-0-10/02/05-15:54:21, Copyright (C) 1999-2004 Digium. Zaptel 1.0.92 dmesg : Zapata Telephony Interface Registered on major 196 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Registered tone zone 0 (United States / North America) Regards, Chris Chris Miller President - Rocket Scientist ScratchSpace Inc. (831) 621-7928 http://www.scratchspace.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
On Oct 27, 2005, at 10:25 AM, Jared Armstrong wrote: I had to turn on the aggressive echo cancellation in the zaptel drivers for mine. Which is much better, but we still get occasional pops. The funny part is only the asterisk side of the connection hears the echo. If you have bad echo problems, we just put a new echo canceler (or rather updates to an old one) into CVS. It's based on KB1, and it's called MG2. It might improve your performance. Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SPA3000 as trunk - no caller ID
FWIW, I've noticed on v3.1.7g that after dialing (via the spa3k), about one of three attempts will cause the pstn line to drop. Not sure as yet what the problem is, but the spa3k did not do that before upgrading firmware. Here is my version Software Version: 3.1.5(GWb) Hardware Version: 2.0.1(42a8) I had mentioned this before, I am downloading 3.1.7 right now. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Daragon Sent: Thursday, October 27, 2005 3:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID Kerry Garrison wrote: During a PSTN call the status screen correctly displays the caller ID information. Well, if the SPA-3000 is picking up the CID, and PSTN CID as VOIP CID is set, and the caller ID isn't being passed to Asterisk, it looks as if the SIP INVITE is being passed to Asterisk before the CID has been detected. But you've obviously thought of that - hence the delay... It may be worth firing up ethereal to check that the CID really isn't in the INVITE. Are you using version 3.1.7 of the Sipura firmware? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF recognition unreliable or absent, ringing lost on 2nd half of 3-way call.
These appear to be a common problems, but after spending half a day reading the wiki and list archives I have not gained much useful knowledge beyond the fact that these are a common problems. I am hoping for some suggestions or pointers to further info. I have an ivr in my incoming context that does a background() and... well, it is an ivr, no need to explain that, I guess. So, testing locally, it works wonderfully. Testing through my DID, provided by IPKall, it is decidedly hit-or-miss. The digits seem to be either not recognized at all or recognized incorrectly better than half the time. Most often, I get the invalid extension playback that I have assigned to the i,1 exten. For a while, I had two test extensions, one 2000 and one 2001. Dialing 2001 usually sent me to 2000 instead. What is making it hard for me to debug is that it *sometimes* works, recognizing the extension I dialed correctly. My peer entry in sip.conf for IPKall contains dtmfmode=rfc2833 as per their recommendation. I have tried setting relaxedtmf=yes in the general section, with no noticeable change. I turned it off again, since the problem seems to be too much relaxation in any case. Looking at the console, I dial 7056 and it sees 7055, I dial 7056 again and it sees 75, I dial 7056 a third time and it sees 706, etc. Seems random and all over the place. Packet loss and/or ordering? Aside from the dtmf issue, incoming calls on the DID work fine and sound excellent. Another issue that may or may not be related, but that I would like to solve, is that when I flash the switch to initiate a three-way call and dial a number, when I flash back to the original call the ringing on the second call stops. I just hear silence until the call connects. When the call does connect, I can send no dtmf at all to whatever is at the other end. To put it another way... You call me. I want to play you a message on my voicemail, say at the office. I flash the hook and get a dial tone, dial my work VM number, and the call starts ringing. I flash back, and the ringing stops. We listen to the silence together until the VM system picks up, but at that point neither of us can send dtmf to log in. (The call works normally otherwise, audio in both directions, etc.) I am sure the answers to these questions require only a basic understanding of the way signaling and bridging work over and across the different technologies, but I am having a really hard time acquiring that understanding. I would be grateful for any help. lyd ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
If you do an fxotune and all of the coefficients are 0, does this mean that fxotune is not making any changes? Based on what Matt has mentioned previously, fxotune only sets the impedence to proper values today. He has not implemented the code to set the coefficients as yet, therefore the expected values are zero's. Are these settings persistent across reboots? The README for fxotune seems to mention that you need to do a fxotune -s in order to reload the card with the analyzed settings (rather than take the 20 minutes it seems to take on my 6 lines). However, if fxotune.conf is all 0's, I sure hope that the settings are persistent on the board! :) The results of fxotune is written to /etc/fxotune.conf; I don't believe they are read back in unless you build something into a bootup script. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
[EMAIL PROTECTED] has the Zaptel from head. Just need to update the zaptel drivers from CVS head you don't have to upgrade the asterisk. Matthew Fredrickson wrote: On Oct 27, 2005, at 12:02 AM, Mark Quitoriano wrote: i tried doing the instruction from voip-info[1] anyway here's my comment with that instruction. when i tried doing /usr/src/zaptel/fxotune -i 4 it gives me this Tuning module 1Failure! Tuning module 2Failure! Tuning module 3Failure! Tuning module 4Failure! how can i debug this? i look at my /var/log/messages and it gives me many of this line Oct 26 17:36:06 sloan kernel: — Set echo registers successfully Oct 26 17:36:25 sloan kernel: — Setting echo registers: I'm using AAH 1.5 stock no modifications. I think they're using versions 1.0.9 for all(e.g. asterisk, zaptel). Don't use 1.0.9. It's old. Try the version of fxotune from head. Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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: Zaptel stop hangs server
[EMAIL PROTECTED] wrote on 10/27/2005 10:20:05 AM: I'll give it a shot. Do you compile it with Zaptel running or diasable it and reboot first? Either way should work fine... Of course you will hang one more time trying to unload the current zaptel drivers. Bill ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] fxotune fails with valid TDM/FXO card
The recent suggestion on the list was to not use 1.0.9 zaptel Chris Miller wrote: Fxotune doesn't appear to work with the latest TDM boards. I have a TDM400P rev I card and receive the following when running fxotune : # ./fxotune -i 4 Tuning module 1 Skipping non-TDM / non-FXO Failure! Tuning module 2 Skipping non-TDM / non-FXO Failure! I didn't see anything obvious in the code that ties this to the card revision, but I recalled seeing something on the list about previous changes in this regard. Any suggestions on getting this to work? System details : Fedora Core 4 Kernel 2.6.13-1.1526_FC4smp Asterisk CVS-v1-0-10/02/05-15:54:21, Copyright (C) 1999-2004 Digium. Zaptel 1.0.92 dmesg : Zapata Telephony Interface Registered on major 196 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXO (FCC mode) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules) Registered tone zone 0 (United States / North America) Regards, Chris Chris Miller President - Rocket Scientist ScratchSpace Inc. (831) 621-7928 http://www.scratchspace.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is it possible to generate or play some white noise in Asterisk?
Is it possible to play or generate some white noise, down an Asterisk call? Some calls I am making are terminating if there is an RTP timeout. Is there some file I can play during the call to fix this? /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip not working suddenly
Anyone know what's causing this: -- SIP read from x.x.x.x:56800: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKba5c00f818387D67 From: Xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B To: sip:[EMAIL PROTECTED];user=phone;tag=as57402fc2 CSeq: 1 ACK Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' lou01*CLI -- SIP read from x.x.x.x:56800: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0 From: Xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B To: sip:[EMAIL PROTECTED];user=phone CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username=user1, realm=asterisk, nonce=07b9f9a3, uri=sip:[EMAIL PROTECTED]:5060;user=phone, response=a8f005540682f07a88e023d50135cce0, algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1130439113 1130439113 IN IP4 192.168.200.16 s=Polycom IP Phone c=IN IP4 192.168.200.16 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Reliably Transmitting (NAT) to x.x.x.x:56800: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0;received=x.x.x.x;rport=568 00 From: Xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B To: sip:[EMAIL PROTECTED];user=phone;tag=as71adaedb Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=56bff437 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from x.x.x.x:56800: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0 From: xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B To: sip:[EMAIL PROTECTED];user=phone CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username=user1, realm=asterisk, nonce=07b9f9a3, uri=sip:[EMAIL PROTECTED]:5060;user=phone, response=a8f005540682f07a88e023d50135cce0, algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1130439113 1130439113 IN IP4 192.168.200.16 s=Polycom IP Phone c=IN IP4 192.168.200.16 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines)--- Ignoring this INVITE request Transmitting (NAT) to x.x.x.x:56800: SIP/2.0 488 Not Acceptable Here (codec error) Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0;received=x.x.x.x;rport=568 00 From: xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B To: sip:[EMAIL PROTECTED];user=phone;tag=as71adaedb Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] please recommend phones with adsi.
On 10/27/05, Chris Coulthurst [EMAIL PROTECTED] wrote: I sure like my Aastra 390, the voicemail ADSI app works pretty well (only a couple of incompleted functions, like not exiting by hanging up the speakerphone, rather than go to a reorder tone. As for the 'look at the wiki' comment, I'm not trying to get on anyone's badside, but Dmitry was asking for recommendations, not documentation. Sorry, not pointing fingers, but I see that 'blanket answer' of going to to Wiki all too often on here lately ;) I hope you are talking about the same post from Dmitry. The following is Dmitrys post: Hello, can somebody recommend me any hard or may be even softphones which support ADSI. I would like to work with Asterisk voicemail application using ADSI. Thanks, Dmitry For which I responded look at the wiki and I even included a direct link to the wiki. I don't see in any way how this doesn't answer his question. In any case it sure doesn't look like he has any clue of *any* ADSI phones, since he asking for a softphone that supports ADSI, while I can't say it doens't exist or there is no use for it (BTW, I'm almost sure it doesn't exist), there sure isn't a market for it. Since ADSI is something made to work on Analog networks. While if you were asking for good jazz music the Sam Goody answer wouldn't do, if you had no clue what jazz music is then the Sam Goody answer is the right answer. In most cases when you see that blanket answer of go to the wiki, it is becuase the person posting the question has thru the question told everyone I havn't seen the wiki yet. Which BTW was the case here. Hope this helps you understand why that answer was in place. Please don't take this as being on my badside I'm just trying to explain to you what RTFM means. All of us are busy with somethings, we take our time to answer the questions here on the list it doesn't mean that we are here to do the work for you so that you could be a lazy bum. If someone is lacking the knowledge of searching the wiki and shows that thru posting that question of any soft phones supporting ADSI, I answerd the question with the most repect I could gather for the 2 seconds by directing them to the wiki, since with that question they showed they had no clue the wiki exists, and if Dmitry will tell me that he did know about the wiki and still posted the question the way he did, then he did not deserve my 2 seconds. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Is it possible to generate or play some white noise in Asterisk?
First you could adjust the rtp timeout (I think its in sip.conf), second the white noise (or CNG, Cofort Noise Gen) is something that was added to the bug tracker not too long ago, although only as an application right now, but if more ppl test it and report (even just that it works) it will push it further, I cant remember the bug number and don't have the time right now to search for it, but it was done by the same person that fixed up that MOH shouldn't use the incoming stream as the timer so that VAD connections still have nice MOH. This hould help you search for it. On 10/27/05, Obelix [EMAIL PROTECTED] wrote: Is it possible to play or generate some white noise, down an Asterisk call? Some calls I am making are terminating if there is an RTP timeout. Is there some file I can play during the call to fix this? /Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Test after Hurricane Wilma
What a creative way to test. GL. On 10/27/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Hi guys. Please disregard this. I'm testing connectivity after being down due to Hurricane Wilma. Thanks, Waldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asteriks configuration
Hi, Is there any possibility to 'point' a FWD, Callware or IPKall number/virtual number to an Asteriks server bypassing the PSTN network to connect cu Asteriks? I need to do a setup like this: 1 local VoIP provider for calls within the country. Calls will be made directly from the endpoint. Another VoIP provider for international calls. The users will call the Asteriks server, get dial tone the dial the international number. Can you point me to some documentation to make this task. And the last question: Is possible to dial into Asteriks server via virtual numbers? Cheers, -- Dazzle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Words for the Asterisk community to live by.
I was sitting at my buddies house, and noticed a little sign that he has on his desk, and thought, these are great words for the Asterisk community to live by. Service Policy: We provide service which is CHEAP, FAST PERFECT. You can only have two. If you want it CHEAP and FAST, It won't be PERFECT If you want it CHEAP and PERFECT, It won't be FAST If you want it FAST and PERFECT, It won't be CHEAP. -- Leif Madsen - http://www.leifmadsen.com http://www.asteriskdocs.org -- Co-Founder http://www.oreilly.com/catalog/asterisk -- Co-Author ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is anyone using OpenSer - A fork of SER?
Folks! I want to know if anyone in the list is using OpenSER, which appears to be a fork of SER. If so can you post Your comments on its functionality? The location where this is available is here: http://openser.org/index.php#about Some of the the features I am impressed with being... 1)Programming command syntax, which was not available in SER. 2)Modular Architecture like Asterisk A list of modules available are here: http://openser.org/diffs-0.9.0.php Seshu Kanuri NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
[EMAIL PROTECTED] wrote on 10/27/2005 03:11:11 PM: Are these settings persistent across reboots? The README for fxotune seems to mention that you need to do a fxotune -s in order to reload the card with the analyzed settings (rather than take the 20 minutes it seems to take on my 6 lines). However, if fxotune.conf is all 0's, I sure hope that the settings are persistent on the board! :) The results of fxotune is written to /etc/fxotune.conf; I don't believe they are read back in unless you build something into a bootup script. Correct. From the README: It will write a configuration file to /etc/fxotune.conf. You will need to have your system run fxotune with the -s flag (`fxotune -s`) to set the module with the previously discovered values from fxotune.conf for it to take affect, so essentially if each time you reboot the machine you need to run `fxotune -s`. You might consider putting it in your startup scripts some time after the module loads and before asterisk runs. However, my fxotune.conf contains only 0's for all 8 of each of 6 lines. I'm wondering does that mean that fxotune had no effect, or that whatever effect it does have is A) Persistent within the card between reboots and B) Not reflected by a fxotune.conf filled with 0's... Tim Massey ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
On Oct 27, 2005, at 3:35 PM, [EMAIL PROTECTED] wrote: However, my fxotune.conf contains only 0's for all 8 of each of 6 lines. I'm wondering does that mean that fxotune had no effect, or that whatever effect it does have is A) Persistent within the card between reboots and B) Not reflected by a fxotune.conf filled with 0's... If you ran it from an older version of the utility (older than a month I would say) I would re-run it just in case. Also, even for all 0's, you want it to set it at boot. I'm not sure that I trust that all of the registers that this effects are initialized to 0 by default. Matthew Fredrickson ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delay ReInvite
Hi all, this is probably a asterisk-devel question but I'll try it here first. Is there a way to delay a ReInvite on SIP? I have an issue where my provider's server is only ~1 ms RTT away and for about 1/3 of the incoming calls I get a 482 Loop Detected error because the ReInvite is processed by the calling server before the ACK packet on Answer(). Asterisk ReInvites right after answering and both packets leave my end virtually simultaneously (within 0.1 ms based on time stamp in tcpdump). I looked at the code in CVS and the 482 Loop Detected message is sent back when an Invite comes in with a call ID of a pending outgoing invite that has not yet been answered (at least this is how I understand it) -- and in this case it would be a loop. It's not a major issue because the calling end re-tries the Invite in a second and then it usually works. This is quite reproducible on my setup and I can provide tcpdump captures if anyone has some ideas. Luki ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 using tdm400p has echo
Are these settings persistent across reboots? The README for fxotune seems to mention that you need to do a fxotune -s in order to reload the card with the analyzed settings (rather than take the 20 minutes it seems to take on my 6 lines). However, if fxotune.conf is all 0's, I sure hope that the settings are persistent on the board! :) The results of fxotune is written to /etc/fxotune.conf; I don't believe they are read back in unless you build something into a bootup script. Correct. From the README: It will write a configuration file to /etc/fxotune.conf. You will need to have your system run fxotune with the -s flag (`fxotune -s`) to set the module with the previously discovered values from fxotune.conf for it to take affect, so essentially if each time you reboot the machine you need to run `fxotune -s`. You might consider putting it in your startup scripts some time after the module loads and before asterisk runs. However, my fxotune.conf contains only 0's for all 8 of each of 6 lines. I'm wondering does that mean that fxotune had no effect, or that whatever effect it does have is A) Persistent within the card between reboots and B) Not reflected by a fxotune.conf filled with 0's... When wctdm is loaded, the registers in the chipsets on the TDM card get initialized. That implies whatever was there is overwritten. The default register values have been preprogrammed into the driver for some time, and those defaults are those applicable to the US telephone lines. So, if you're using a TDM card in the US, the fxotune functions provided today don't really add any value to the operation of the card since those values are already programmed (eg, impedence). I'm assuming (but haven't bothered to look) that if the card is used in a non-US location, the loadzone parameter in /etc/zaptel.conf changes the impedence setting to what is appropriate for that country. If that is case, then the current value of running fxotune is apparently zero. About a year or so ago, I played around with changing the coefficients manually and did not find where those changes had any significant impact on audio quality (including echo). But, there has been many changes and improvements within zaptel and the canceller where maybe playing with the coefficients might now be noticed. The echo issue seems to always come back to the software canceller and how well it performs in various environments. Those that have attempted to optimize the canceller have indicated its previous operating range is rather limited when compared to dedicated cancellers. I only know enough about it to know that I don't have the knowledge or background to offer improvements, but I do understand the issues. Matt has been working on canceller improvements including the MG2 that he posted today. So, gut feeling (as of this moment anyway) is that fxotune is not the answer to echo. The changes being made involving the canceller have had very noticable improvements starting with the KB canceller. (Now off to play with MG2. :) Rich ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Realtime sip register=
Juan Salas wrote: yes, I tested too and it's works. The Problem is when we want to add (or delete) registers without reload the asterisk. We are using it like a border server wich is registered like many users in a SER machine and the real endpoints are registered in the asterisk. I guess one could create a manager interface function for adding registry entries while running. The app itself has to make sure that these are also stored in a config file, so that they still exist after reload. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Delay ReInvite
Luki wrote: Hi all, this is probably a asterisk-devel question but I'll try it here first. Is there a way to delay a ReInvite on SIP? I have an issue where my provider's server is only ~1 ms RTT away and for about 1/3 of the incoming calls I get a 482 Loop Detected error because the ReInvite is processed by the calling server before the ACK packet on Answer(). Asterisk ReInvites right after answering and both packets leave my end virtually simultaneously (within 0.1 ms based on time stamp in tcpdump). I looked at the code in CVS and the 482 Loop Detected message is sent back when an Invite comes in with a call ID of a pending outgoing invite that has not yet been answered (at least this is how I understand it) -- and in this case it would be a loop. It's not a major issue because the calling end re-tries the Invite in a second and then it usually works. This is quite reproducible on my setup and I can provide tcpdump captures if anyone has some ideas. Which version of Asterisk are you using? I have a vague memory of fixing this in CVS head, but I might be wrong. Can't really check here, sorry. Test with CVS head, and if you still have problems please open a bug report in the bug tracker at bugs.digium.com with the call trace. THank you. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] UK BT IDSN30e 'pass through' withTE205P/AvayaArgentOffice?
The argent office does not support DASS2 so I suspect your circuit will be ISDN30e anyway. Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: 26 October 2005 15:38 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [Asterisk-Users] UK BT IDSN30e 'pass through' withTE205P/AvayaArgentOffice? On Wed, Oct 26, 2005 at 03:33:16PM +0100, Mark Ackroyd wrote: You should also ensure the PRI is really configured for EuroISDN, many BT PRI's are actually UK ISDN which Asterisk doesn't support (it's an older version). I had a problem along these lines, when I first started with asterisk, the PRI was originally DASS2, but needed to be Q931 Full ETSI for it to work. In between we had it configured for Q931 1/2 ETSI and the outbound didn't work. What the actually difference is I don't know. BT uses mainly Marconi System X exchanges, these will by default configure PRI lines as ISDN v85 (I seem to remember) while EuroISDN is v110 (or something similar). Steve -- NetTek Ltd Fax +44-(0)20 7483 2455 Skype / In stevekennedyuk / UK +442088167166 / US +13106518226 Vonage UK +442079932612 / US +13108577715 / UK mob 07775 755503 Personal Blog http://stevekennedy.blogspot.com Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream GXP-2000
We're having a rather serious echo problemusing the Grandstream GXP-2000's with Asterisk 1.0.9. I'm wondering if there is something I'm overlooking that might be an easy fix. The echo seems to be worst on internal SIP to SIP calls but you do get it every once in a while on outgoing calls through the PRI. It's not the speakerphone echo problem, we're running the 1.0.1.12 firmware that pretty much fixes that. It seems like most of the echo cancellation functions are for outgoing calls through the phone company. Is this a more likely a phone problem? We've got about 50 of these phones all doing the same thing. -- | Erick Baum ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ANNOUNCEMENT : A2Billing - AreskiCC V3 new release
Hi I´ve just installed a2billing using PHP Version 5.0.4, MySQL version 4.1.12 and Asterisk CVS-v1-0-06/27/05, verified database installation and can see webpage, login, create cards, etc, but I cant hear anything when I call the extension: extension.conf ; use 6608600 as access number to enter the calling card system exten = 6608600,1,Answer exten = 6608600,2,Wait,2 exten = 6608600,3,DeadAGI(a2billing.php|1) exten = 6608600,4,Wait,2 exten = 6608600,5,Hangup DEBUG:: (level2 in a2billing.conf) *CLI *CLI *CLI *CLI -- Executing Answer(SIP/264-ce26, ) in new stack -- Executing Wait(SIP/264-ce26, 2) in new stack -- Executing DeadAGI(SIP/264-ce26, a2billing.php|1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: IDCONFIG : 1 a2billing.php|1: a2billing.php|1: A2Billing AGI internal configuration: a2billing.php|1: Array a2billing.php|1: ( a2billing.php|1: [debug] = 2 a2billing.php|1: [logger_enable] = 1 a2billing.php|1: [log_file] = /tmp/a2billing.log a2billing.php|1: [setlanguage_deprecate] = 1 a2billing.php|1: [say_goodbye] = a2billing.php|1: [play_menulanguage] = a2billing.php|1: [force_language] = EN a2billing.php|1: [intro_prompt] = a2billing.php|1: [len_cardnumber] = 10 a2billing.php|1: [len_voucher] = 15 a2billing.php|1: [min_credit_2call] = 0 a2billing.php|1: [use_dnid] = a2billing.php|1: [no_auth_dnid] = Array a2billing.php|1: ( a2billing.php|1: [0] = 2400 a2billing.php|1: [1] = 2300 a2billing.php|1: ) a2billing.php|1: a2billing.php|1: [number_try] = 3 a2billing.php|1: [say_balance_after_auth] = 1 a2billing.php|1: [say_balance_after_call] = a2billing.php|1: [say_timetocall] = 1 a2billing.php|1: [cid_enable] = 1 a2billing.php|1: [cid_askpincode_ifnot_callerid] = 1 a2billing.php|1: [cid_auto_create_card] = a2billing.php|1: [cid_auto_create_card_typepaid] = POSTPAY a2billing.php|1: [cid_auto_create_card_credit] = 0 a2billing.php|1: [cid_auto_create_card_credit_limit] = 1000 a2billing.php|1: [cid_auto_create_card_tariffgroup] = 6 a2billing.php|1: [sip_iax_friends] = a2billing.php|1: [sip_iax_pstn_direct_call_prefix] = 9 a2billing.php|1: [sip_iax_pstn_direct_call] = a2billing.php|1: [dialcommand_param] = |30|HL(%timeout%:61000:3) a2billing.php|1: [dialcommand_param_sipiax_friend] = |30|HL(360:61000:3) a2billing.php|1: [switchdialcommand] = a2billing.php|1: [record_call] = a2billing.php|1: [monitor_formatfile] = gsm a2billing.php|1: [base_currency] = usd a2billing.php|1: [agi_force_currency] = a2billing.php|1: [currency_association] = Array a2billing.php|1: ( a2billing.php|1: [0] = usd:prepaid-dollar a2billing.php|1: [1] = mxn:pesos a2billing.php|1: [2] = eur:euro a2billing.php|1: [3] = all:credit a2billing.php|1: ) a2billing.php|1: a2billing.php|1: [file_conf_enter_destination] = prepaid-enter-dest a2billing.php|1: [file_conf_enter_menulang] = prepaid-menulang2 a2billing.php|1: [debugshell] = 0 a2billing.php|1: [currency_association_internal] = Array a2billing.php|1: ( a2billing.php|1: [usd] = prepaid-dollar a2billing.php|1: [mxn] = pesos a2billing.php|1: [eur] = euro a2billing.php|1: [all] = credit a2billing.php|1: ) a2billing.php|1: a2billing.php|1: ) a2billing.php|1: a2billing.php|1: AGI Request: a2billing.php|1: Array a2billing.php|1: ( a2billing.php|1: [agi_request] = a2billing.php a2billing.php|1: [agi_channel] = SIP/264-ce26 a2billing.php|1: [agi_language] = en a2billing.php|1: [agi_type] = SIP a2billing.php|1: [agi_uniqueid] = 1130449297.8 a2billing.php|1: [agi_callerid] = Rafo a2billing.php|1: [agi_dnid] = 6608600 a2billing.php|1: [agi_rdnis] = unknown a2billing.php|1: [agi_context] = internos a2billing.php|1: [agi_extension] = 6608600 a2billing.php|1: [agi_priority] = 3 a2billing.php|1: [agi_enhanced] = 0.0 a2billing.php|1: [agi_accountcode] = a2billing.php|1: ) a2billing.php|1: a2billing.php|1: 264 ; SIP/264-ce26 ; 1130449297.8 ; ; 6608600 -- AGI Script a2billing.php completed, returning 0 -- Executing Wait(SIP/264-ce26, 2) in new stack -- Executing Hangup(SIP/264-ce26, ) in new stack == Spawn extension (internos, 6608600, 5) exited non-zero on 'SIP/264-ce26' *CLI *CLI what I am doing wrong? thanks rafael ,On 10/26/05, Areski K [EMAIL PROTECTED] wrote: Dear Friends,Great day for the callingcard-lovers !!!I am pleased to release the version 3.0 of AreskiCC !!!http://www.areski.net/a2billing/ http://www.voip-info.org/wiki/view/A2Billing Little unexpected change, we got a new name... bit more serious A2BillingMany many features have been added, lot bugs solved and a bunch of goodenhancements made!The key newest features : - Full MYSQL support- USE PHP-AGI LIB 2.14- CALLERID SUPPORT AUTHENTICATION- MUSICONHOLD CUSTOMIZATION BY DIALPREFIX- UPLOADING TOOLS TO CONFIGURE MUSICONHOLD- INVOICES PDF / HTML - ADD NET REPORTING FROM ASTERISK-STAT# calls compare# monthly traffic# daily load- DEFAULT DIALING FOR RATECARD- FAILOVER
[Asterisk-Users] Outgoing fax detect
I need to detect when a fax machine answeres an outgoing call. NV_FaxDetect and the zaptel fax detect seem to only work in calls originated FROM a fax machine, not for calls ANSWERED by a fax. Thanks -- Larry Host NuWorld Telecom, Ltd. 858-334-9355 Cell tfbunm AOL and Yahoo IM [EMAIL PROTECTED] MSN IM ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
