[Asterisk-Users] Meetme2 - web interface not working
I got the asterisk-part working, when i join a conference, i can see the entry created in my pgsql database using webmin. it looks like: --- Table meetme_user in database meetme user_id confno chan_name fd ztc_chanztc_confno ztc_confmodeflag 6 50 Zap/2-1 17 2 1023772 0 - but when enter conference room 50 in web interface (did setup defines.php of course) - i always get No user in this conference room. - no error or whatsoe apache 2 PHP Version 4.3.3 register_globals = on Asterisk 1.0.2-BRIstuffed-0.2.0-RC2 What can i try? Thanks Jens ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can someone tell me what is going on from this debug?
On Fri, 12 Nov 2004 15:50:13 -0500 (EST), Doug Eubanks [EMAIL PROTECTED] wrote: Can someone tell me why Asterisk is sending 404 instead of passing this call to the demo? I have replaced the IPs with descriptions This is the actual asterisk debug, Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 0x1(G723) Found peer 'sip.simflex.net' Looking for 19995551212 in default Reliably Transmitting (no NAT): SIP/2.0 404 Not Found It would appear you do not have 19995551212 as a valid extension in your default context Regards Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AgentCallBackLogin
W licie z pi, 12-11-2004, godz. 23:55, Shawn Dillon pisze: I have the AgentCallBackLogin working well when the support technician logs into the queue manually. If there a way to get certain extensions to automatically log into the queue? That way I do not have to worry about help desk staff forgetting to log into the support queue and never receiving support calls. As far as I know, it's enought to add those extensions as a members in queue.conf. -- Graf0 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Authenticate or DISA?
On Fri, 12 Nov 2004 18:32:55 -0700, Paul Fielding [EMAIL PROTECTED] wrote: I want to authenticate to the phone system, then be able to call an extension or dial an outside line. My preferred method would be to use DISA, because a) it's non-verbal - ie. it doesn't talk, just provides dialtone, and b) it provides dialtone. My alternative seems to be to use Authenticate, and upon authenticating simply send the caller to the appropriate context to punch in extensions or calls. The problem with this is a) it voices the authentication - ie please enter password which to me is inviting people to try to figure it out, and b) after authenticating you don't get a dialtone, just silence. After the Authenticte why not do a Playtones(Dial) this will give dialtone Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple Question
On Sun, 14 Nov 2004 16:44:12 -, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Is this quite simple to set up and can I attach asterix to my landline via a standard modem? Yes no go to http://www.voip-info.org/wiki-Asterisk and read learn try and read try agin Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip relay with asterisk
I've got the following setup: SIP Client -- SER -- Asterisk -- Iptel.org SIP account I'm now trying to place an outgoing call, which has to be authenticated at the iptel.org proxy server (which ser can't do) but I seem to be getting 407 packets with the IP of the asterisk machine as realm. SIP ClientSER * Iptel.org Invite --- Invite - Invite --- --- 407 realm ip_of_* Afaik the realm iptel.org should provide to * should be iptel.org instead of the IP of the * box (which would indicate an error at iptel?) What I'm trying to archive is that the * box authenticated the calls to iptel and then leave the call alone (so I will have to find out how to get * out of the media path) I'm still new to all this, but I think this could work. Kind regards, E. Versaevel Extensions.conf [sip_in_from_carrier] exten = _XX, 1, Dial(SIP/[EMAIL PROTECTED],20,r) ;Not a 10 digit number exten = s,1,Answer exten = s,2,MusicOnHold() exten = s,3,Hangup ;Timeout exten = t,1,Answer exten = t,2,Background(conf-invalid) ;exten = t,3,MusicOnHold() exten = t,4,Hangup ;Hangup exten = h,1,Hangup [sip_in_from_ser] exten = _., 1, Dial(SIP/[EMAIL PROTECTED],20,r) ;Not a 10 digit number exten = s,1,Answer exten = s,2,MusicOnHold() exten = s,3,Hangup ;Timeout exten = t,1,Answer exten = t,2,Background(pin-invalid) ;exten = t,3,MusicOnHold() exten = t,4,Hangup ;Hangup exten = h,1,Hangup [default] exten = s, 1, Background(conf-invalid) exten = s, 2, Hangup Sip.conf [general] port=5065 disallow=all allow=ulaw register = asterisk:[EMAIL PROTECTED] ;Incomming from ser register = iptel:[EMAIL PROTECTED]/iptel_alias ;Incomming from iptel [sip.carrier] type=user realm=iptel.org username=iptel secret=iptel host=sip.iptel.org canreinvite=no context=sip_in_from_carrier [sip.carrier] type=peer host=sip.iptel.org context=sip_in_from_carrier [sip.ser] type=user realm=sermachine host=sermachine canreinvite=no context=sip_in_from_ser [sip.ser] type=peer host=sermachine context=sip_in_from_ser ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN - Asterisk - PSTN Call quality
Hi there, Having some issues with call quality when taking calls from E1, using Asterisk to reroute the call back out onto E1. Sometimes theres quite a big echo and others the line is just very scratchy. Call quality for incoming calls to VoIP is fine. To redirect the incoming call I use an AGI that fires off the Dial command to redial the extension back out over PSTN. Is this the right way to redirect a call out over PSTN? Would doing this cause any kind of call quality loss? Cheers for any suggestions or help, Ben Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] maximum retries error
Hi all, I have two xlite clients which are attempting to make a call through asterisk. The call seems to connect and the clients are both marked as connected on either side how ever no audio is transmitted. One client is behind nat (the asterisk server is also behind nat). I am getting the following error and would really really appreciate if someone could help me sorting out what the issue is. I have included my config files below: Thanks as always, Aisling. *CLI Nov 15 11:09:52 WARNING[13999]: chan_sip.c:665 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) ;sip.conf [general] port = 5060 ; Port to bind to (SIP bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all ; Allow all codecs allow=gsm allow=alaw allow=ulaw ;XLite client on my laptop [2000] type=friend ; This device takes and makes calls username=2000 ; Username on device secret=suzuki ; Password for device host=dynamic ; This host is not on the same IP addr every time mailbox=2000 ;regexten=2000 nat=yes ;auth=md5 context=sip callerid=Aisling2000 ;dmtfmode=rfc2833 canreinvite=no ;qualify=8000 ;reinvite=no disallow=all allow=gsm allow=ulaw allow=alaw ;qualify=500 ; Activate the message waiting light if this ; voicemailbox has messages in it ;xlite client on juliens laptop [2003] type=friend ; This device takes and makes calls username=2003 ; Username on device secret=2003 ; Password for device host=dynamic ; This host is not on the same IP addr every time mailbox=2003 ;regexten=2003 nat=yes ;auth=md5 context=sip callerid=mum2003 ;dmtfmode=rfc2833 canreinvite=no ;qualify=8000 disallow=all allow=gsm allow=ulaw allow=alaw ;qualify=500 [2004] type=friend ; This device takes and makes calls username=2004 ; Username on device secret=2004 ; Password for device host=dynamic ; This host is not on the same IP addr every time mailbox=2004 ;regexten=2004 nat=yes ;auth=md5 context=sip callerid=mum2004 ;dmtfmode=rfc2833 canreinvite=no ;qualify=8000 disallow=all allow=gsm allow=ulaw allow=alaw ;qualify=500 [2005] type=friend ; This device takes and makes calls username=2005 ; Username on device secret=2005 ; Password for device host=dynamic ; This host is not on the same IP addr every time mailbox=2005 ;regexten=2005 nat=yes ;auth=md5 context=sip callerid=mum2005 ;dmtfmode=rfc2833 canreinvite=no ;qualify=8000 disallow=all allow=gsm allow=ulaw allow=alaw ;qualify=500 ;XLite client on Juliens laptop [2001] type=friend ; This device takes and makes calls username=2001 ; Username on device secret=bla; Password for device host=dynamic ; This host is not on the same IP addr every time mailbox=2001 ;regexten=2001 nat=yes ;auth=md5 context=sip callerid=Julien2001 ;dmtfmode=rfc2833 canreinvite=no ;qualify=8000 ;reinvite=no disallow=all allow=gsm allow=ulaw allow=alaw ;qualify=500 [2006] type=friend ; This device takes and makes calls username=2006 ; Username on device secret=2006 ; Password for device host=dynamic ; This host is not on the same IP addr every time mailbox=2006 ;regexten=2006 nat=yes ;auth=md5 context=sip callerid=whatever2006 ;dmtfmode=rfc2833 canreinvite=no ;qualify=8000 ;reinvite=no disallow=all allow=gsm allow=ulaw allow=alaw ;qualify=500 ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ResponseTimeout problem
On Sun, 2004-11-14 at 22:13 -0700, Joseph wrote: [snip] Yes, I was looking at it already but it is available in ver. 1.0.0 and up; I'm on 0.9 on Gentoo. Gentoo is kind of slow when it comes to Asterisk. There is an unstable ver. 1.0.2 in unstable branch but it doesn't compile (there is an error when compiling). FYI: the 1.0.2 version is actually stable and part of the stable branch which can be downloaded with: cvs co -r v1-0 zaptel libpri asterisk The v1-0 tag will get you the latest stable release from cvs while it is also possible to use v1-0-1 to get version 1.0.1, v1-0-2 for 1.0.2 etc. So I will have to learn how to upgrade using CVS or wait for Gentoo stable version. Checkout the shell script that was posted to the mailing list last week. It automates the upgrading building process. You can find it here: http://www.szmidt.org/asterisk/asterisk-update.sh If I use CVS I'm not sure if startup scrip will be upgraded as well in /etc/init.d/ When I install my updated stable-cvs builds (which are rpms), it upgrades/replaces the startup scripts. Not really an issue afaik. Regards, Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax preferred codec question?
hallo, could somebody help me, i would like to select ilbc as preferred codec but dont disable gsm totally, i can only make a call with ilbc if i disable the gsm codec in iax.conf, if i enable gsm and make call to the same enpoint, always the gsm codec is choosen as audio codec. any idea whats wrong here? how to choose a preferred codec without disabling gsm? thanks for help, alex [krmtu] type=friend secret=mypass username=krmtu notransfer=no auth=md5,plaintext,rsa host=dynamic context=default ;allow=all disallow=all allow=ilbc allow=gsm callerid=krmtu1114 mailbox=1114 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN - Asterisk - PSTN Call quality
On Mon, 2004-11-15 at 21:18, Ben Merrills wrote: Hi there, Having some issues with call quality when taking calls from E1, using Asterisk to reroute the call back out onto E1. Sometimes theres quite a big echo and others the line is just very scratchy. Call quality for incoming calls to VoIP is fine. To redirect the incoming call I use an AGI that fires off the Dial command to redial the extension back out over PSTN. Is this the right way to redirect a call out over PSTN? Would doing this cause any kind of call quality loss? I had (still have) the same problem. So far, I've found two possible solutions: a) Get a new motherboard b) Get a new motherboard and a new TE410p (instead of my current TE405p) I am still suffering this problem since I can't really afford to buy a new motherboard, cpu, memory besides, how do I know that after buying all that, it will even solve the problem? Looking forward to a motherboard whitelist ? someone? Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PSTN - Asterisk - PSTN Call quality
Hmm, When we first got our TE410p (we have 2 at the minute) it was in a normal IDE ATA100 box with a P4, and the static on the line was really, REALLY bad. We don't have this issue now we use a Compaq with SCSI, unless we reroute the call back onto PSTN. It doesn't always happen I might add though! It seems to be some landlines that have more problems than others. Mobiles tend to be fine. Is this a Digital - Analogue issue? Strange that it should be fine the rest of the time :( Yes, a hardware guide for Asterisk would be a god send! I'm getting two new DELLs next week to play with (SCSI again). I'll let you know how well they work with Asterisk! Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Goryachev Sent: 15 November 2004 12:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PSTN - Asterisk - PSTN Call quality On Mon, 2004-11-15 at 21:18, Ben Merrills wrote: Hi there, Having some issues with call quality when taking calls from E1, using Asterisk to reroute the call back out onto E1. Sometimes there's quite a big echo and others the line is just very scratchy. Call quality for incoming calls to VoIP is fine. To redirect the incoming call I use an AGI that fires off the Dial command to redial the extension back out over PSTN. Is this the right way to redirect a call out over PSTN? Would doing this cause any kind of call quality loss? I had (still have) the same problem. So far, I've found two possible solutions: a) Get a new motherboard b) Get a new motherboard and a new TE410p (instead of my current TE405p) I am still suffering this problem since I can't really afford to buy a new motherboard, cpu, memory besides, how do I know that after buying all that, it will even solve the problem? Looking forward to a motherboard whitelist ? someone? Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: zap channel won't send/receive calls
Hi, I have a fxo card configured in my asterisk pbx.. but cant seem to make calls from POTS --- sip or from sip pots when I place a call, asterisk seems to begin the call processing -- Executing Dial(SIP/fahmy-452d, Zap/1/4168880686) in new stack -- Called 1/4168880686 -- Zap/1-1 answered SIP/fahmy-452d and thats where it stops... no dial tone, the other side is not ringing, and nothing is happening on the line finally when I hang up this is what I see on console -- Hungup 'Zap/1-1' == Spawn extension (from-sip-internal, 94168880686, 1) exited non-zero on 'SIP/fahmy-452d' -- Executing Hangup(SIP/fahmy-452d, ) in new stack == Spawn extension (from-sip-internal, h, 1) exited non-zero on 'SIP/fahmy-452d' and if I make a call into my asterisk box from an external number, the call is detected, and a few seconds later it is hung up... still no dial tone -- Starting simple switch on 'Zap/1-1' -- Executing PrivacyManager(Zap/1-1, ) in new stack -- CallerID Present: Skipping -- Executing Dial(Zap/1-1, SIP/1100SIP/1400|30) in new stack Nov 14 20:32:45 WARNING[1632]: chan_sip.c:1386 create_addr: No such host: 1100 Nov 14 20:32:45 NOTICE[1632]: app_dial.c:743 dial_exec: Unable to create channel of type 'SIP' Nov 14 20:32:45 WARNING[1632]: chan_sip.c:1386 create_addr: No such host: 1400 Nov 14 20:32:45 NOTICE[1632]: app_dial.c:743 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time -- Executing VoiceMail2(Zap/1-1, b1100) in new stack -- Playing 'vm-theperson' (language 'en') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/local/1100/INBOX/msg format: wav, 0x8107580 and even though voicemail is being played i can't seem to hear it, then the call gets disconnected before I have a chance to leave voicemail I've tried everything and this is my last resort... any help is greatly appreciated here is my dmesg output... sorry for the long email dmesg output: -- Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 12 for device 00:09.0 PCI: Sharing IRQ 12 with 00:10.1 wcfxo: DAA mode is 'FCC' Found a Wildcard FXO: Generic Clone ISDN subsystem Rev: 1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1/1.1.4.1 loaded HiSax: Linux Driver for passive ISDN cards HiSax: Version 3.5 (module) HiSax: Layer1 Revision 1.1.4.1 HiSax: Layer2 Revision 1.1.4.1 HiSax: TeiMgr Revision 1.1.4.1 HiSax: Layer3 Revision 1.1.4.1 HiSax: LinkLayer Revision 1.1.4.1 HiSax: Approval certification failed because of HiSax: unauthorized source code changes /etc/zaptel.conf fxsks = 1 loadzone = us defaultzone = us /etc/asterisk/zapata.conf --- [channels] language=en context=from-analog signalling=fxs_ks usecallerid=yes echocancel=yes echocancelwhenbridged=yes channel = 1 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] skinny error
check to make sure you have a ip address added to teh skinny.conf file.. if your even using skinny. Jason On Sun, 14 Nov 2004 10:42:08 +0200, Thomas Andrews [EMAIL PROTECTED] wrote: What does this error mean: Nov 14 10:35:12 WARNING[24733]: Unable to get our IP address, Skinny disabled I looked in channels/chan_skinny.c and it looks like ourhost[] is never initialised ? $ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] skinny error
On Mon, Nov 15, 2004 at 07:32:29AM -0500, Jason p wrote: check to make sure you have a ip address added to teh skinny.conf file.. if your even using skinny. Yup, that's it. Thanks Jason. Regards, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] re: DVG-1120
They work fine Asterisk via MGCP (If is a 1120M, which is what ATT uses). This device is a little unique in that it is a router/firewall and ATA in one. The MGCP/RTP only uses the external interface. Make sure set the NAT Adress in the 1120 for you Lan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Sunday, November 14, 2004 2:25 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] re: DVG-1120 Hello, I know the d-link units (DVG-1120 ATA and their router as well) are supposed to work well with asterisk...does anyone know if the units that come with ATT callvantage are locked, or can they be used w/asterisk or SER? And if they are locked, is it linksys no way out locking or a simple password thing? thanks, yair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] skinny error
Hi Thomas, On Sun, 14 Nov 2004 10:42:08 +0200, Thomas Andrews [EMAIL PROTECTED] wrote: What does this error mean: Nov 14 10:35:12 WARNING[24733]: Unable to get our IP address, Skinny disabled I have had problems when the IP address of the Asterisk host has not been explicitly defined on the line bind = x.x.x.x in the file skinny.conf. In older versions of code bind = 0.0.0.0 was sufficient. I now find that you must indicate the actual IP address of the LAN card on the Asterisk server or skinny support will not startup correctly. HTH Darren -- Darren Storer Comgate Telco|Internet|Broadcast ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk nagios plugin
hi I've written, or upgraded a little more, a plugin for asterisk/nagios, just in case someone should be interested. it uses the manager interface to connect and checks staus. it's a dirty hack, but it works. see https://sourceforge.net/tracker/? func=detailaid=746083group_id=29880atid=541465 for more info roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Music On Hold Problem
Hi What codec are you using? Best to use iLBC, 711U/A caused the same problem with our system. What handsets are you using? Grandstream work well with iLBC firmware ver.11. The problem is that there are not to many phones that work well with iLBC. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Venu V Sent: Sunday, November 14, 2004 3:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Music On Hold Problem Recently I have configured Music On Hold option in asterisk PBX. But I am unable to listen to the audio properly and morever its getting breaks for every 3 seconds. If any one know about this. Please help me Thanks Regards V.Venu -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, November 14, 2004 12:07 PM To: [EMAIL PROTECTED] Subject: Asterisk-Users Digest, Vol 4, Issue 181 Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: SysMaster and GPL Violation (Brian) 2. Re: getting callerid from spa3k to asterisk (Randy Bush) 3. my asterisk drops connection when remote side putsme on hold? (Steve Prior) 4. Cisco ATA and G729 (kido noagbodji) 5. Remote answer not detected (DB) 6. Re: SysMaster and GPL Violation (Voip Business) 7. RE: Cable for T1 connection: Crossover or straightthrough? (Franceen Thompson) 8. RE: Cisco ATA and G729 (Franceen Thompson) 9. Queue/AgentCallbackLogin Problems (Franceen Thompson) -- Message: 1 Date: Sat, 13 Nov 2004 19:30:06 -0700 From: Brian [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SysMaster and GPL Violation To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Are you saying that those of us that are using the product should not be allowed to voice our opinions about its licensing, development and maintenance? That we should all just shut up and take whatever Mark co. give us? If that's the case, then this is most definitely NOT an open-source project at all. -Original Message- From Brandon Patterson Sent: Saturday, November 13, 2004 7:15 PM Uh ok...So when will Asterisk be a licensed product? Will it take the form of a Redhat sort of platform... Fedora with Redhat the pay me money side of the house? Just a simple question: When can we expect to see Asterisk the licensed as in paid for version ? Brandon Right now. As far as I know, you just need to contact Digium's sales department and negotiate a licensing agreement with them. -- Message: 2 Date: Sat, 13 Nov 2004 19:11:10 -0800 From: Randy Bush [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: getting callerid from spa3k to asterisk To: splatters [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii if i have two sip contexts for my spa3k, on inbound and one outbound, e.g. [spa3k-in] type=friend host=dynamic port=5061 auth=md5 secret=pfui qualify=1000 canreinvite=yes context=ext-in42 [spa3k-out] type=peer auth=md5 secret=pfui username=outpass fromuser=outpass host=spa3k.bogus.com port=5061 nat=no canreinvite=yes context=ext-in42 and the spa3k's PSTN / Subscriber Information / User ID: = spack-in, the incoming connection from spa3k to * is being routed to the spa3k-out context, not the spa3-in context. see appended. i suspect this is a bug in * 1.0.1. i found the problem, or at least a work-around. if i reverse the order of the above two sip contexts, the incoming call is properly routed to the spa3k-in sip context as opposed to the wrong one, spa3k-out. my guess is that * is traversing a list and taking the first context which has the ip address and port it wants without checking the context name against the name which was received over the wire. so it depends on what order the contexts are inserted in the list. aii! randy -- Message: 3 Date: Sat, 13 Nov 2004 22:33:58 -0500 From: Steve Prior [EMAIL PROTECTED] Subject: [Asterisk-Users] my asterisk drops connection when remote side puts me on hold? To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed I've
Re: [Asterisk-Users] AU FreeBSD PRI Hardware
On Mon, 2004-11-15 at 07:46, Talbot Neil wrote: Hi, I was wondering if there is any PRI hardware that is Austel certified and works well with Asterisk under FreeBSD??? If anyone has any information please let me know as I seem to be having problems finding any documentation in regards to this. Isn't that the matter of finding a .au distributor/reseller, that did the certification ? As far as i've understood it, it works the way, that once somebody imports a card, they get it certified and put their stikker on it. This however doesn't mean, that you can import the same card in parallel and use it, as it allready is certified. Every company that imports telephony hardware has to certify it against Austel and they probably will not allow their competitors to use their certification. (Correct me, if i'm wrong). Kind regards, Martin List-Petersen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk
After a recent upgrade to asterisk HEAD, my asterisk startup scripts don't properly start asterisk. They have since May, which is the last time I upgraded. I am on Slackware 9.1, running kernel 2.4.26. After reboot, lsmod shows wct1xxp, then zaptel, which would indicate it now loads out of order? Shouldn't zaptel be loaded first? Maybe my original install is a little hacked. Where do you load all your modules and asterisk from on startup of your server? I have a T100P and a TDM400P installed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] xlite and asterisk
Ashling O'Driscoll wrote: I am also getting a call not approved error on xlite??I know a fw people have also come across this problem because Ive seen threads posted on it but the solution has never been posted. If anyone has idea please let me know. Kindest regards, Aisling. Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] xlite and asterisk Date: Thu, 11 Nov 2004 13:06:52 -0600 X-Lite works fine for me with plain text passwords. Unlike the stuff below, though, I'm not using nat=yes. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chad Scott Sent: Thursday, November 11, 2004 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] xlite and asterisk It's been awhile since I've played with X-Lite, but I think it absolutely *has* to use the MD5 auth stuff. Use md5secret rather than secret in sip.conf. You'll have to MD5 hash your password... there's documentation on this in the Wiki. -Chad On Nov 10, 2004, at 9:25 AM, Ashling O'Driscoll wrote: Hi, Hope somebody can help. I have two xlite clients that register with asterisk. They are called 2000 and 2001. 1)When 2000 rings 2001 a '404 not found' message is returned even though he is registered with asterisk. 2)When 2001 rings 2000, a 'call not approved' error is returned. I found a thread regarding the 'call not approved' error in the asterisk archives but no solution was posted. I have included the relevant portion of my config files below. If any further info is needed please let me know. Also how is it possible to dial a sip address e.g. sip:[EMAIL PROTECTED] from an xlite client? Thanks again, Aisling. sip.conf ;xlite client 1 [2000] type=friend username=2000 secret=whatever nat=yes host=dynamic mailbox=100 [2001] type=friend username=2001 secret=bla nat=yes host=dynamic mailbox=101 extensions.conf exten =3D 2000,1,Dial(SIP/2000,20) exten =3D 2001,1,Dial(SIP/2001,20) ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. - - - - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. - --- ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of
Re: [Asterisk-Users] skinny error
Hi Darren, On Mon, Nov 15, 2004 at 02:21:27PM +, Storer, Darren wrote: In older versions of code bind = 0.0.0.0 was sufficient. I now find that you must indicate the actual IP address of the LAN card on the Asterisk server or skinny support will not startup correctly. That's exactly what I found. I just put the IP of the ethernet card in there and the error went away. Regards, Thomas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme2 - web interface not working
On Mon, 2004-11-15 at 08:45, Jens Hansen wrote: I got the asterisk-part working, when i join a conference, i can see the entry created in my pgsql database using webmin. it looks like: --- Table meetme_user in database meetme user_id confno chan_name fd ztc_chanztc_confno ztc_confmode flag 6 50 Zap/2-1 17 2 1023772 0 - but when enter conference room 50 in web interface (did setup defines.php of course) - i always get No user in this conference room. - no error or whatsoe apache 2 PHP Version 4.3.3 register_globals = on Asterisk 1.0.2-BRIstuffed-0.2.0-RC2 What can i try? I would check the connection between your asterisk and the database. I've got MeetMe2 running on the exact same version, just with mysql instead of postgresql and saw the same thing in the beginning. In the end MeetMe2 just never got a connect to the database due to a typo in the configuration. Check if MeetMe2 is sending queries at all to the SQL server. Kind regards, Martin List-Petersen Dublin, Eire ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AU FreeBSD PRI Hardware
Martin List-Petersen wrote: On Mon, 2004-11-15 at 07:46, Talbot Neil wrote: Hi, I was wondering if there is any PRI hardware that is Austel certified and works well with Asterisk under FreeBSD??? If anyone has any information please let me know as I seem to be having problems finding any documentation in regards to this. Isn't that the matter of finding a .au distributor/reseller, that did the certification ? As far as i've understood it, it works the way, that once somebody imports a card, they get it certified and put their stikker on it. This however doesn't mean, that you can import the same card in parallel and use it, as it allready is certified. Every company that imports telephony hardware has to certify it against Austel and they probably will not allow their competitors to use their certification. (Correct me, if i'm wrong). That is roughly the deal in most places, so it probably is in .au It is not unreasonable, either. The body holding the certification is responsible for ensuring all units shipped comply with the certification. Even if they wanted to, they couldn't do that for parallel imports. It used to be that factory inspections and other horrible complexity (otherwise known as free holiday trips to Asia :-) ) were required in many places. Mostly that has been dropped. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple options to Dial command - what is the correct format?
I am needing to have multiple options to the Dial() command. I tried: exten 1212,1,Dial(SIP/333,20,D(123)A(beep)) but it did not work. What format do I use for having multiple options for the Dial command. THanks, jerry ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] manager api: how to handle failed calls
Hello Nicolas, first of all I want to thank you. You are the first guy
give me an answer. I already posted this issue two times but nobody was
interested in it.
I tried your sugggestion but it doesn't work. In the mean time I
upgraded to v1.0.2 but things remain the same or even worse ( I have in
the log a new channel name OutgoingSpoolFailed but the behaviour is the
same).
Any way perhaps I found a solution and I would like to ask what do you
think about it:
This is the request:
/bin/echo "Action: Originate";\
/bin/echo "Channel: Local/[EMAIL PROTECTED]";\
/bin/echo "Variable:
callid=123456|number=X|url="">
/bin/echo "Context: chiamamezzi3";\
/bin/echo "Exten: s";\
/bin/echo "Priority: 1";\
/bin/echo "Callerid: Asterisk Automatic Wardial";\
/bin/echo "Timeout: 1";\
/bin/echo "Async: True";\ ** this is important otherwise fail
extension doesn't work **
/bin/echo "ActionId: 10";\
This is the context
[chiamamezzi3]
exten = _.X,1,Dial(ZAP/g1/${EXTEN})
exten = _.X,2,NoOp( _.X DIALSTATUS is ${DIALSTATUS}, number is
${number} )
exten = _.X,3,SetGlobalVar(status=${DIALSTATUS})
exten = s,1,Wait,1
exten = s,2,Answer
exten = s,3,DigitTimeout,5
exten = s,4,ResponseTimeout,10
exten = s,5,system(/prd/asterisk/log.sh "Variabili:
[menuid=${menuid}] [number=${number}] [redirectnum=${redirectnum}]
[url="" [callid=${callid}]")
exten = s,6,Goto(chiamamezzi-${menuid},s,1)
exten = t,1,Goto(noanswer,s,1)
exten = failed,1,NoOp( failed DIALSTATUS is ${status} number is
${number})
exten = failed,2,Hangup()
exten = h,1,Hangup()
As I told you, If the call is ok dialplan goes on s, 1
If the call fails I proceed on _.X,2 and 3 and I need to set a
globalvariable to let to handle it in the failed extension.
It works but I don't know if it could work on multi request environment
. It would be much better to use a channel variable or even better, to
have DIALSTATUS directly available .
Regards
Luca Casavola
Nicols Gudio wrote:
Hello,
Comments inline..
The question is how to correctly handle failed calls.
In my application I want to make hundreds of outgoing calls automatically.
When the callee pick up the phone he gets a playback message and give an
acknowledge by means of dtmf code.
I make use of manager command originate, something like
Action:originate
channel: ZAP/g1/
Variable:X|Y|Z
extension: test
the extension test is something like
[test]
exten s,1 , wait ()
exten s, 2 , answer ()
exten s, 3 playback(XX)
The problem is since I don't use the application dial inside the extension
I cannot get any value from
DIALSTATUS or HANGUPCAUSE variable
I tried several strategies:
1)
change the logic and use local pseudo channel
In the originate command if I use channel: local/[EMAIL PROTECTED]/n
where test1 is:
[test1]
exten = _.,1,Dial(ZAP/g1/g${EXTEN})
exten = _.,2,NoOp( 2 HANGUPCAUSE is ${HANGUPCAUSE})
exten = _.,3,NoOp( 2 DIALSTATUS is ${DIALSTATUS})
exten = _.,4,NoOp( number is ${number})
exten = _.,5,Hangup
I got the correct HANGUP value ( ie BUSY) but unfortunately I cannot see
the variables set on the originate command.
I wonder why not?
Maybe, (just maybe, I did not try it myself) the originate variables
are passed using asterisk CVS-HEAD and variable names prefixed with
underscore... Eg: Use variable _X instead of X in the originate
command. Let me know if it works.
Regards,
___
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Re: [Asterisk-Users] Asterisk
i load them in /etc/rc.local - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, November 15, 2004 9:47 AM Subject: [Asterisk-Users] Asterisk After a recent upgrade to asterisk HEAD, my asterisk startup scripts don't properly start asterisk. They have since May, which is the last time I upgraded. I am on Slackware 9.1, running kernel 2.4.26. After reboot, lsmod shows wct1xxp, then zaptel, which would indicate it now loads out of order? Shouldn't zaptel be loaded first? Maybe my original install is a little hacked. Where do you load all your modules and asterisk from on startup of your server? I have a T100P and a TDM400P installed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer # - Intermittent with Cisco 7905 SIP Phone
We are having some problems with our Cisco 7905 phones (SIP 7.x) and the transfer # functionality not working intermittently. Has anyone else experienced this? Problem Description: User dials # to transfer a call to another SIP extension. Both the user and the party on the other end hear a dial tone when # is dialed but Asterisk does not prompt for extension to transfer to. Intermittent problem, most of the time dialing # to transfer works fine. Regards, J. Staalenburg Teksavers, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
What is the general consensus on the Polycom SIP Phones? I am getting random gargled up sounds on mine and I really do think it is the Polycom Regards, Michael DiMartino ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk
After a recent upgrade to asterisk HEAD, my asterisk startup scripts don't properly start asterisk. They have since May, which is the last time I upgraded. I am on Slackware 9.1, running kernel 2.4.26. After reboot, lsmod shows wct1xxp, then zaptel, which would indicate it now loads out of order? Shouldn't zaptel be loaded first? Maybe my original install is a little hacked. Where do you load all your modules and asterisk from on startup of your server? I have a T100P and a TDM400P installed. I'm not a slackware user, however on RH linux I had noticed some of those same issues. Two items seem to be at the root: 1. the wcfxs module was renamed to wctdm some time ago, but a normal cvs checkout doesn't handle that unless you do a 'make config' from within the zaptel directory, or you manually edit the startup script changes. The wctdm is needed for the TDM card. 2. On one newly installed system with a T100P and TDM400P installed, I had to change the load order of the drivers (within the startup script) so that the TDM400P (wxtdm) got loaded _before_ the T100P driver. The digium folks seem to think that was because the T100P has never actually been configured (which we aren't ready to do at this time). But, changing the order fixed our problem. Translate into slackware scripts as needed Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA and G729
Hello all, * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or threekeys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? Funny enough when i disabled the gsm, g729, g723 codec, it works fine (no need to pressany key), with the alaw and the ulaw codec. I guess the ATA default to the alaw and the ulaw when it does not find the other codecs, However that is alot of bandwidth i am wasting ... Before i licensed the g729 codec, is there a way i can "test" it? Many Thanks Kido - Original Message - From: kido noagbodji To: [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 3:36 AM Subject: [Asterisk-Users] Cisco ATA and G729 Hi all, I am new to asterisk. I was able, but not without pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work with the PBX. Three remarks: * On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In the h323.conf file i enabled those codec. Everything works great!!! * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or threekeys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? *Also when i set my ATA codec to g729 and in asterisk i allow=g729, i get a very low weird sound. What is that due to? I am guessing that i don't have the codec installed on the system. Is there an open source g729 codec available for FreeBSD? Any help will be very much appreciated, Thanks. Kido ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NETDEV WATCHGOG eth0 timeout
I have run into a very peculiar problem. I have installed Fedora Core 3 on an Athlon64 System (MSI K8Neo Platinum motherboard and an AMD64-3000 cpu). This system has an onboard Gigabit ethernet which is identified by the kernel and operates correctly. I installed the latest asterisk for use as a SIP only server / gateway. There are 5 cisco 7940 IP Phones and codec g729 is used, and a cisco router 3640 is used as a gateway to the pbx via its isdn bri voice lines. One of the problems is that when i try to call a line to the pbx, the network on the linux machines stops almost completely. In the logs (message and dmesg) there are entries NETDEV WATCHDOG: eth0 timeout after a minute the network is restored. This happens only when a place a call to pass thru asterisk to the pbx via the cisco gateway. Also if i put the ethernet in promiscuous mode, it does not hangup. I have not run into a behavior like this one before with 32bit systems (Intel / AMD). Is it a problem of X86_64 or with the onboard ethernet controller? Why it is done only with asterisk? For example FTP works ok with heavy load... Any help will be highly appreciated. Savas Pavlidis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA and G729
kido noagbodji wrote: Hello all, * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? Funny enough when i disabled the gsm, g729, g723 codec, it works fine (no need to press any key), with the alaw and the ulaw codec. I guess the ATA default to the alaw and the ulaw when it does not find the other codecs, However that is a lot of bandwidth i am wasting ... Before i licensed the g729 codec, is there a way i can test it? Many Thanks Kido - Original Message - *From:* kido noagbodji mailto:[EMAIL PROTECTED] *To:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Sent:* Sunday, November 14, 2004 3:36 AM *Subject:* [Asterisk-Users] Cisco ATA and G729 Hi all, I am new to asterisk. I was able, but not without pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work with the PBX. Three remarks: * On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In the h323.conf file i enabled those codec. Everything works great!!! * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? * Also when i set my ATA codec to g729 and in asterisk i allow=g729, i get a very low weird sound. What is that due to? I am guessing that i don't have the codec installed on the system. Is there an open source g729 codec available for FreeBSD? Any help will be very much appreciated, Thanks. Kido ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users $10 for a license from digium Iqbal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA and G729
You can only use g729 in pass-thru mode without paying for the licensing fees. G729 is probably the best codec around. If you plan on having any sort of thriving business based on VoIP, g729 would be the way to go. I don't suggest PCMU or PCMA for production. The ATA will pass a list of supported codecs to the Asterisk server and based on what you have allowed in your h323.conf or sip.conf file, that will be what codec is selected. Your audio quality problems could also be traced to a problem with transcoding between different codecs (i.e alaw - ulaw problem). I suggest you try one by one, all the codecs available to you and disallow/allow codecs in your configuration until you can find the source of your problem. On Monday 15 November 2004 03:23 pm, kido noagbodji wrote: Hello all, * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? Funny enough when i disabled the gsm, g729, g723 codec, it works fine (no need to press any key), with the alaw and the ulaw codec. I guess the ATA default to the alaw and the ulaw when it does not find the other codecs, However that is a lot of bandwidth i am wasting ... Before i licensed the g729 codec, is there a way i can test it? Many Thanks Kido - Original Message - From: kido noagbodji To: [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 3:36 AM Subject: [Asterisk-Users] Cisco ATA and G729 Hi all, I am new to asterisk. I was able, but not without pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work with the PBX. Three remarks: * On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In the h323.conf file i enabled those codec. Everything works great!!! * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? * Also when i set my ATA codec to g729 and in asterisk i allow=g729, i get a very low weird sound. What is that due to? I am guessing that i don't have the codec installed on the system. Is there an open source g729 codec available for FreeBSD? Any help will be very much appreciated, Thanks. Kido --- --- ___ 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 -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Where can I find searchable version of this list?
Thanks, robert ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where can I find searchable version of this list?
On Mon, 2004-11-15 at 10:34 -0500, Augustyn, Robert non Unisys wrote: www.google.com, add site:lists.digium.com to the search terms and it will limit it to the list archives. BTW, you should include the question as the body of the message even if it is the same as the subject. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] xlite and asterisk
I am also getting a call not approved error on xlite??I know a fw people have also come across this problem because Ive seen threads posted on it but the solution has never been posted. If anyone has idea please let me know. We ran into this problem too and we found that the settings for Asterisk under SIP Settings SIP Proxy HAS to be in the Default category. Putting it in any other category caused the phone to register properly but yields Call not approved when you dial out. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SendText
On Sun, 2004-11-14 at 13:02, Alessandro Gatti wrote: Hello, I was trying to use SendText to send a message to an extension, but it seems as if the message is being sent to the caller instead of the callee... e.g.: exten = 123, 1, SendText(hello world) Does anyone have any suggestion on how to override the behavior? Many thanks, Alex Well, like most applications it performs on the channel that called it. That means the caller in the terms you used. So when you dial extension 123 in your example the SendText() application will send hello world to you since you are the channel that executed it. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 trunking - timing - ztdummy??
Hello! Perhaps someone can spread i little bit light on this: I want to trunk two Asterisk systems with each other. System A, behind a NAT-Firewall and System B with a real IP address. aix.conf on B: [mytrunk] host=dynamic username=mytrunk auth=md5 secret=yyy trunk=yes iax.conf on A: register = mytrunk:[EMAIL PROTECTED] When I make a reload an B I get the following: Nov 15 16:32:32 WARNING[-1244329040]: chan_iax2.c:6427 build_peer: Unable to support trunking on peer 'mytrunk' without zaptel timing I have downloaded the zaptel package, compiled it ( including ztdummy, which may be what I need ) and installed it. The kernel modules load: ztdummy 3492 0 zaptel228996 1 ztdummy crc_ccitt 2176 1 zaptel I don't know how to configure zaptel ( /etc/zaptel.conf ) to get this to work. I have no hardware, I only want timing for the IAX2 trunk ( and later on for Conference calls ). I have also read about the rtc package but have not tried it. I may have overseen very basic things... Please enlighten me! Regards:Håkan pgpQHZNKq9tFm.pgp Description: PGP signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where can I find searchable version of this list?
Thanks, robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Monday, November 15, 2004 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Where can I find searchable version of this list? On Mon, 2004-11-15 at 10:34 -0500, Augustyn, Robert non Unisys wrote: www.google.com, add site:lists.digium.com to the search terms and it will limit it to the list archives. BTW, you should include the question as the body of the message even if it is the same as the subject. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AgentCallBackLogin
I have the AgentCallBackLogin working well when the support technician logs into the queue manually. If there a way to get certain extensions to automatically log into the queue? That way I do not have to worry about help desk staff forgetting to log into the support queue and never receiving support calls.As far as I know, it's enought to add those extensions as a members inqueue.conf.-- Graf0 Ok, I will share more details with my particular installation. 1) In my extensions.conf I have the following; exten = 997,1,AgentCallBackLogin(999|[EMAIL PROTECTED]) 2) In my agents.conf I have the following group=1 agent = 999,1234,Test Agent 3) In my queues.conf I have he following member = Agent/999 When an agent dials extension 997 and enters their password they will then be included in the support queue for calls. If they do not call extension 997 and enter their password when a call is placed into the support queue a message appears on the console stating that no one is answering the support queue. Is there a way to get a certain extension to automatically log into a support queue? Or do I need to have every technician , at the start of every shift, log into the support queue manually? As an aside , this community has been very helpful in getting my Asterisk box up and running. Thanks to all. Shawn Dillon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FXO setup
Hi, Re: X100P Asterisk does not hangup automatically after caller leave a voicemail message and hangup.! Asterisk does not hangup automatically after the caller hangup in the Auto attendant menu system! What variables should I change to have * automatically hangup if the caller hangup? Right now, I have a variable set to a maximum of 60 seconds to hangup. All comments are greatly appreciated. Darly ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MYSQL Dialplan Question
I am new to Asterisk, and I am having trouble connecting to the MySQL database located on the same machine as my Asterisk box. When the dialplan tried to connect to MySQL database, I get the following error message on the Asterisk console. Nov 15 09:29:41 WARNING[39760]: pbx.c:1279 pbx_extension_helper: No application 'MYSQL' for extension (default, s, 5) Here is the corresponding line in my dialplan. exten = s,5,MYSQL(Connect connid localhost admin abc123 asteriskdb) I am able to connect to the database with the same username and password using the MySQL console. Am I missing something in my installation or configuration that would cause this? Thanks, Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transferring calls from a Zyxel P2000w
Hello every body!
I'm having problems with a Zyxel P2000W phone, it looks i'm unable to transfer
call from this phone.
The entry for the phone in sip.conf is this one:
[andy1]
type=friend
username=andy1
host=dynamic
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
context=sip
In extensions.conf i have this:
exten = 206,1,Dial(SIP/andy1,40,trT)
exten = 206,2,Voicemail(u${EXTEN})
exten = 206,3,Hangup
exten = 206,4,Voicemail(b${EXTEN})
exten = 206,5,Hangup
On the phone i have the firmware version WJ.00.0a
With the G729 codec.(Yes, i tried G711.u and G711.a)
Asterisk version: Asterisk CVS-HEAD-07/27/04
I have cisco 7960 in my system as well but with those i can transfer calls, so
my question now is, is it a configuration issue in my asterisk or is it just
that the phone limitation?
Thanks in advance,
Esteban
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Re: [Asterisk-Users] Cisco ATA and G729
Hi, Thanks Brian. As you said the cisco tries the codec one by one. When i only enable codec that can be supported i have no sound problem. Thanks. Question though, As you suggested i would like to use the g729 codec but before i purchase it from digium can i have a sort of demo version? Also do i have an easy way to install the codec under FreeBSD? It was tough enough to install asterisk even with the FreeBSD ports. BTW for the $10 per channel should i consider $10 for H323 channel $10 for the SIP channel (for instance), or is it $10 per number of concurrent calls wanted regardless of the protocols used? Thanks, Kido - Original Message - From: Brian Wilkins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 15, 2004 10:35 AM Subject: Re: [Asterisk-Users] Cisco ATA and G729 You can only use g729 in pass-thru mode without paying for the licensing fees. G729 is probably the best codec around. If you plan on having any sort of thriving business based on VoIP, g729 would be the way to go. I don't suggest PCMU or PCMA for production. The ATA will pass a list of supported codecs to the Asterisk server and based on what you have allowed in your h323.conf or sip.conf file, that will be what codec is selected. Your audio quality problems could also be traced to a problem with transcoding between different codecs (i.e alaw - ulaw problem). I suggest you try one by one, all the codecs available to you and disallow/allow codecs in your configuration until you can find the source of your problem. On Monday 15 November 2004 03:23 pm, kido noagbodji wrote: Hello all, * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? Funny enough when i disabled the gsm, g729, g723 codec, it works fine (no need to press any key), with the alaw and the ulaw codec. I guess the ATA default to the alaw and the ulaw when it does not find the other codecs, However that is a lot of bandwidth i am wasting ... Before i licensed the g729 codec, is there a way i can test it? Many Thanks Kido - Original Message - From: kido noagbodji To: [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 3:36 AM Subject: [Asterisk-Users] Cisco ATA and G729 Hi all, I am new to asterisk. I was able, but not without pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work with the PBX. Three remarks: * On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In the h323.conf file i enabled those codec. Everything works great!!! * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? * Also when i set my ATA codec to g729 and in asterisk i allow=g729, i get a very low weird sound. What is that due to? I am guessing that i don't have the codec installed on the system. Is there an open source g729 codec available for FreeBSD? Any help will be very much appreciated, Thanks. Kido -- - --- ___ 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 -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MYSQL Dialplan Question
On Mon, 2004-11-15 at 10:04 -0600, Shaun Tierney wrote: I am new to Asterisk, and I am having trouble connecting to the MySQL database located on the same machine as my Asterisk box. When the dialplan tried to connect to MySQL database, I get the following error message on the Asterisk console. Nov 15 09:29:41 WARNING[39760]: pbx.c:1279 pbx_extension_helper: No application 'MYSQL' for extension (default, s, 5) Here is the corresponding line in my dialplan. exten = s,5,MYSQL(Connect connid localhost admin abc123 asteriskdb) I am able to connect to the database with the same username and password using the MySQL console. Am I missing something in my installation or configuration that would cause this? Yes you are missing something in your installation. From reading the error message I can see you did not build the mysql application. My guess would be that it is not enabled in the makefile. Go edit it, compile, install, and try again. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO setup
Re: X100P Asterisk does not hangup automatically after caller leave a voicemail message and hangup.! Asterisk does not hangup automatically after the caller hangup in the Auto attendant menu system! What variables should I change to have * automatically hangup if the caller hangup? Right now, I have a variable set to a maximum of 60 seconds to hangup. All comments are greatly appreciated. Could be that you've not got the x100p configured correctly to detect the hangup; can't tell from what you've posted (twice). Assuming you actually get some sort of hangup notification from your telco, I'd look in zapata.conf samples for various ways to address it. Since we don't have a clue what country you're in, etc, no one is going to be able to help you. For voicemail, you might try maxsilence=10. Not a clue as to how to address your auto-attendant issue since you've not provide us with any configuration data whatsoever. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TMD400 FXO - Nokia 32 GSM (Hangup Problems)
Hi, I have a TDM400 FXO module connected to a Nokia 32 GSM Terminal ( http://www.nokia.com/nokia/0,,56025,00.html ). Outgoing calls from asterisk to the Nokia work flawlessly. Incoming calls from the Nokia are working fine when asterisk hangs up the call. However, when the far-end hangs up (i.e., the Nokia GSM hangs up the call), asterisk detects the hangup but fails to put the line back onhook and the GSM terminal stays with the line busy until I reload the zaptel/wcfxs modules. I think this is due to some kind of electrical problem and did some voltage measurements by using the debug mode in the wcfxs module. As I understand it, the voltage should be around 48V when the line is onhook. When a call comes in it starts to drop until it reaches 7V, meaning the call is connected. During the call is stays at this value and when the Nokia hangs up it drops the battery for 500ms (this process is called Disconnect Supervision). Asterisk detects this correctly as the Hangup signal and should open the circuit so that voltage goes back to 48V indicating the line is onhook and ready for another call. The problem is that the voltage only goes to ~37V instead of 48V and the Nokia terminal still thinks Asterisk has the line offhook. I have included some of these logs. Any help/hints on what causes this problem will be greatly appreciated. I've also looked at the source code of wcfxs.c and despite having a broad idea of how it works, I'm not at all comfortable with messing around with it's low level internals. Hints on any hacks to the code that could solve this would be great. Oh, and can anyone tell me what Debounce is/does ? Thanks, Leandro -- /var/log/messages --NOTE: 48V meaning line is onhook and ready for a call Nov 13 19:27:21 raider kernel: Module 2: Installed -- AUTO FXO (FRANCE mode) Nov 13 19:27:21 raider kernel: ProSLIC on module 3, product 0, version 0 Nov 13 19:27:21 raider kernel: Module 3: Not installed Nov 13 19:27:21 raider kernel: Found a Wildcard TDM: Wildcard TDM400P REV H (4 modules) Nov 13 19:27:21 raider kernel: Card 3: Voltage: 48 Debounce 0 Nov 13 19:27:21 raider kernel: 7278595 Polarity reversed (0 - 1) Nov 13 19:27:22 raider kernel: Card 3: Voltage: 49 Debounce 63 Nov 13 19:27:24 raider last message repeated 6 times Nov 13 19:27:24 raider kernel: Card 3: Voltage: 48 Debounce 63 Nov 13 19:27:25 raider kernel: Card 3: Voltage: 49 Debounce 63 Nov 13 19:27:25 raider kernel: Setting FXS hook state to 0 (00) Nov 13 19:27:25 raider kernel: Setting FXS hook state to 0 (00) Nov 13 19:27:25 raider kernel: Registered tone zone 2 (France) Nov 13 19:27:25 raider kernel: Card 3: Voltage: 49 Debounce 63 Nov 13 19:27:27 raider last message repeated 5 times Nov 13 19:27:28 raider kernel: Card 3: Voltage: 48 Debounce 63 Nov 13 19:27:28 raider kernel: Card 3: Voltage: 48 Debounce 63 ... Nov 13 19:42:32 raider last message repeated 47 times Nov 13 19:42:33 raider kernel: Card 3: Voltage: 47 Debounce 63 Nov 13 19:42:33 raider kernel: Card 3: Voltage: 48 Debounce 63 Nov 13 19:42:36 raider last message repeated 6 times --NOTE: This is when the Nokia rings Asterisk Nov 13 19:42:36 raider kernel: Card 3: Voltage: 38 Debounce 63 Nov 13 19:42:36 raider kernel: RING on 2/3! Nov 13 19:42:36 raider kernel: Card 3: Voltage: 38 Debounce 63 Nov 13 19:42:37 raider kernel: Card 3: Voltage: 38 Debounce 63 Nov 13 19:42:37 raider kernel: Card 3: Voltage: 45 Debounce 63 Nov 13 19:42:37 raider kernel: NO RING on 2/3! Nov 13 19:42:38 raider kernel: Card 3: Voltage: 38 Debounce 63 Nov 13 19:42:38 raider kernel: Card 3: Voltage: 38 Debounce 63 Nov 13 19:42:38 raider kernel: Card 3: Voltage: 37 Debounce 63 Nov 13 19:42:39 raider kernel: Card 3: Voltage: 28 Debounce 63 --NOTE: 7V indicates the call is connected Nov 13 19:42:39 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:40 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:40 raider kernel: Card 3: Voltage: 6 Debounce 63 Nov 13 19:42:40 raider kernel: Card 3: Voltage: 9 Debounce 63 Nov 13 19:42:41 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:42 raider last message repeated 3 times Nov 13 19:42:42 raider kernel: Card 3: Voltage: 9 Debounce 63 Nov 13 19:42:43 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:43 raider kernel: Card 3: Voltage: 6 Debounce 63 Nov 13 19:42:44 raider kernel: Card 3: Voltage: 9 Debounce 63 Nov 13 19:42:44 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:44 raider kernel: Card 3: Voltage: 7 Debounce 63 Nov 13 19:42:45 raider kernel: Card 3: Voltage: 9 Debounce 63 --NOTE: At this point Nokia hangs up and drops the battery Nov 13 19:42:45 raider kernel: Battery loss: 2 (63 debounce) Nov 13 19:42:45 raider kernel: Battery loss: 2 (62 debounce) Nov 13 19:42:45 raider kernel: Battery loss: 1 (61 debounce) Nov 13 19:42:45 raider kernel: Battery loss: 1 (60 debounce) Nov 13 19:42:45 raider kernel: Battery loss: 1 (59 debounce) ... Nov 13
RE: [Asterisk-Users] SendText
That makes sense. I will need to figure out how to use it send it to the callee.. Thanks, Alessandro -Original Message- From: Seth Remington [mailto:[EMAIL PROTECTED] Sent: Monday, November 15, 2004 7:51 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SendText On Sun, 2004-11-14 at 13:02, Alessandro Gatti wrote: Hello, I was trying to use SendText to send a message to an extension, but it seems as if the message is being sent to the caller instead of the callee... e.g.: exten = 123, 1, SendText(hello world) Does anyone have any suggestion on how to override the behavior? Many thanks, Alex Well, like most applications it performs on the channel that called it. That means the caller in the terms you used. So when you dial extension 123 in your example the SendText() application will send hello world to you since you are the channel that executed it. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transferring calls from a Zyxel P2000w
I don't believe the phone has the ability to transfer calls, I
remember looking for this and not finding anything.
-Chris
On Mon, 15 Nov 2004 10:06:16 -0600, Esteban Barrientos Abarca
[EMAIL PROTECTED] wrote:
Hello every body!
I'm having problems with a Zyxel P2000W phone, it looks i'm unable to
transfer call from this phone.
The entry for the phone in sip.conf is this one:
[andy1]
type=friend
username=andy1
host=dynamic
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info
context=sip
In extensions.conf i have this:
exten = 206,1,Dial(SIP/andy1,40,trT)
exten = 206,2,Voicemail(u${EXTEN})
exten = 206,3,Hangup
exten = 206,4,Voicemail(b${EXTEN})
exten = 206,5,Hangup
On the phone i have the firmware version WJ.00.0a
With the G729 codec.(Yes, i tried G711.u and G711.a)
Asterisk version: Asterisk CVS-HEAD-07/27/04
I have cisco 7960 in my system as well but with those i can transfer calls,
so my question now is, is it a configuration issue in my asterisk or is it
just that the phone limitation?
Thanks in advance,
Esteban
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Re: [Asterisk-Users] Voicemail shorter then (ex) 2sec - don't accept
On Sun, 2004-11-14 at 13:02, Joseph wrote: In which configuration file I can specify that I don't want to accept messages for example shorter then 2sec. ? I've looked in voicemail.conf but I couldn't find any setting that will support this option. In most cases message shorter then 2 or 3sec will not contain any message and I don't want system to record them and sending an email to me. You were looking in the right config file. The parameter is called maxmessage. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA and G729
Brian Wilkins wrote: You can only use g729 in pass-thru mode without paying for the licensing fees. G729 is probably the best codec around. If you plan on having any sort of thriving business based on VoIP, g729 would be the way to go. I don't suggest PCMU or PCMA for production. The ATA will pass a list of supported codecs to the Asterisk server and based on what you have allowed in your h323.conf or sip.conf file, that will be what codec is selected. Your audio quality problems could also be traced to a problem with transcoding between different codecs (i.e alaw - ulaw problem). I suggest you try one by one, all the codecs available to you and disallow/allow codecs in your configuration until you can find the source of your problem. Humm..How well is G.729 with Music on Hold ? I've used G.711 for some time (and now, some iLBC), but of course, this is a home system, and need to pass the wife test. I would even go G.722 if required :-) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA and G729
The registration code is tied to your MAC address. For instance, I have one E1 card with 30 channels for testing purposes. So I purchased thirty g729 licenses at $10 each. The channels can be resused, so the one time fee is $300. All you need to do to install the codec is download the codec_g729.so file and place it in your modules directory. Then, download the registration program from the Digium website once you receive your registration code. It will tie the registration code to your MAC Address. Go here to download and purchase the codec: http://www.digium.com/index.php?menu=asterisk_g729 The g729 codec is patented, so you must pay for it if you want to use it. On Monday 15 November 2004 04:08 pm, kido noagbodji wrote: Hi, Thanks Brian. As you said the cisco tries the codec one by one. When i only enable codec that can be supported i have no sound problem. Thanks. Question though, As you suggested i would like to use the g729 codec but before i purchase it from digium can i have a sort of demo version? Also do i have an easy way to install the codec under FreeBSD? It was tough enough to install asterisk even with the FreeBSD ports. BTW for the $10 per channel should i consider $10 for H323 channel $10 for the SIP channel (for instance), or is it $10 per number of concurrent calls wanted regardless of the protocols used? Thanks, Kido - Original Message - From: Brian Wilkins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 15, 2004 10:35 AM Subject: Re: [Asterisk-Users] Cisco ATA and G729 You can only use g729 in pass-thru mode without paying for the licensing fees. G729 is probably the best codec around. If you plan on having any sort of thriving business based on VoIP, g729 would be the way to go. I don't suggest PCMU or PCMA for production. The ATA will pass a list of supported codecs to the Asterisk server and based on what you have allowed in your h323.conf or sip.conf file, that will be what codec is selected. Your audio quality problems could also be traced to a problem with transcoding between different codecs (i.e alaw - ulaw problem). I suggest you try one by one, all the codecs available to you and disallow/allow codecs in your configuration until you can find the source of your problem. On Monday 15 November 2004 03:23 pm, kido noagbodji wrote: Hello all, * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? Funny enough when i disabled the gsm, g729, g723 codec, it works fine (no need to press any key), with the alaw and the ulaw codec. I guess the ATA default to the alaw and the ulaw when it does not find the other codecs, However that is a lot of bandwidth i am wasting ... Before i licensed the g729 codec, is there a way i can test it? Many Thanks Kido - Original Message - From: kido noagbodji To: [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 3:36 AM Subject: [Asterisk-Users] Cisco ATA and G729 Hi all, I am new to asterisk. I was able, but not without pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work with the PBX. Three remarks: * On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In the h323.conf file i enabled those codec. Everything works great!!! * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? * Also when i set my ATA codec to g729 and in asterisk i allow=g729, i get a very low weird sound. What is that due to? I am guessing that i don't have the codec installed on the system. Is there an open source g729 codec available for FreeBSD? Any help will be very much appreciated, Thanks. Kido - - - --- ___ 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 -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] Cisco ATA and G729
G729 sounds better than a cell phone to me. There is no noticable difference, the way we use it here, between Asterisk and a regular phone call. On Monday 15 November 2004 04:43 pm, Julio Arruda wrote: Brian Wilkins wrote: You can only use g729 in pass-thru mode without paying for the licensing fees. G729 is probably the best codec around. If you plan on having any sort of thriving business based on VoIP, g729 would be the way to go. I don't suggest PCMU or PCMA for production. The ATA will pass a list of supported codecs to the Asterisk server and based on what you have allowed in your h323.conf or sip.conf file, that will be what codec is selected. Your audio quality problems could also be traced to a problem with transcoding between different codecs (i.e alaw - ulaw problem). I suggest you try one by one, all the codecs available to you and disallow/allow codecs in your configuration until you can find the source of your problem. Humm..How well is G.729 with Music on Hold ? I've used G.711 for some time (and now, some iLBC), but of course, this is a home system, and need to pass the wife test. I would even go G.722 if required :-) ___ 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 -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transferring calls from a Zyxel P2000w
On Mon, 15 Nov 2004 11:25:55 -0500, Chris TenHarmsel [EMAIL PROTECTED] wrote: I don't believe the phone has the ability to transfer calls, I remember looking for this and not finding anything. You need to use # transfer check wiki Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MYSQL Dialplan Question
Well, I looked at the makefile and I could not see any options for SQL. So I did a grep for SQL on the distribution files and found that in version 0.7.0 support for MySQL was removed, so I'm guessing I'm just going to have to switch to Postgres or something. Thanks for the help, Shaun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, November 15, 2004 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MYSQL Dialplan Question On Mon, 2004-11-15 at 10:04 -0600, Shaun Tierney wrote: I am new to Asterisk, and I am having trouble connecting to the MySQL database located on the same machine as my Asterisk box. When the dialplan tried to connect to MySQL database, I get the following error message on the Asterisk console. Nov 15 09:29:41 WARNING[39760]: pbx.c:1279 pbx_extension_helper: No application 'MYSQL' for extension (default, s, 5) Here is the corresponding line in my dialplan. exten = s,5,MYSQL(Connect connid localhost admin abc123 asteriskdb) I am able to connect to the database with the same username and password using the MySQL console. Am I missing something in my installation or configuration that would cause this? Yes you are missing something in your installation. From reading the error message I can see you did not build the mysql application. My guess would be that it is not enabled in the makefile. Go edit it, compile, install, and try again. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MYSQL Dialplan Question
You need to download the asterisk-addons to have mysql support now. It was only moved to its own project due to licensing changes with MySQL. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shaun Tierney Sent: Monday, November 15, 2004 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MYSQL Dialplan Question Well, I looked at the makefile and I could not see any options for SQL. So I did a grep for SQL on the distribution files and found that in version 0.7.0 support for MySQL was removed, so I'm guessing I'm just going to have to switch to Postgres or something. Thanks for the help, Shaun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, November 15, 2004 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MYSQL Dialplan Question On Mon, 2004-11-15 at 10:04 -0600, Shaun Tierney wrote: I am new to Asterisk, and I am having trouble connecting to the MySQL database located on the same machine as my Asterisk box. When the dialplan tried to connect to MySQL database, I get the following error message on the Asterisk console. Nov 15 09:29:41 WARNING[39760]: pbx.c:1279 pbx_extension_helper: No application 'MYSQL' for extension (default, s, 5) Here is the corresponding line in my dialplan. exten = s,5,MYSQL(Connect connid localhost admin abc123 asteriskdb) I am able to connect to the database with the same username and password using the MySQL console. Am I missing something in my installation or configuration that would cause this? Yes you are missing something in your installation. From reading the error message I can see you did not build the mysql application. My guess would be that it is not enabled in the makefile. Go edit it, compile, install, and try again. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transferring calls from a Zyxel P2000w
On Mon, 2004-11-15 at 16:25, Chris TenHarmsel wrote: On Mon, 15 Nov 2004 10:06:16 -0600, Esteban Barrientos Abarca [EMAIL PROTECTED] wrote: Hello every body! I'm having problems with a Zyxel P2000W phone, it looks i'm unable to transfer call from this phone. The entry for the phone in sip.conf is this one: [moved top post to bottom post] I don't believe the phone has the ability to transfer calls, I remember looking for this and not finding anything. -Chris Correct. The ZyXel can only handle one call at a time. However: http://bugs.digium.com/bug_view_page.php?bug_id=0002460 is a effort to fix that problem. Three party call transfers Asterisk managed. Slán lait, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VICIDIAL in windows xp
Title: Mensaje hello everybody.. I´m trying to finallize with astguiclient/vicidial installation with the scratch-install . Everything works great but 2 stuffs I may see now. with astguiclient running on windows xp -- ok -VICIDIAL when I launchc:\AST_VICI\astVICIDIAL_0.8.pl an error appears and vicidial doesn´t launch.. at first perl asked for time::hires so I made ppm install Time::HiRes and installed HiResversion 1.49 So I tryied again but now askes for an newer version not available at active perl. this´s the error... time::hires object version 1.59 does not match $time::hires::XS_version 1.55 at c:/perl/lib/dynaloader.pm line 253 compilation failed in require at vicidial.pl line 83 begin failed--compilation aborted at vicidial.pl line 83 Someone knows how may I fix this? -ASTGUICLIENT - VICIDIAL web administration at welcome page (http://voip.local/astguiclient/welcome.php) there are a couplo of broken links... -http://voip.local/astguiclient/lists.php -http://voip.local/astguiclient/campaigns.php I didn´t found the pages.. anyone knows? Thanks to all... --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 01/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AgentCallBackLogin
In queues.conf, try: member = Local/extension@extension-context So, for instance, something like: member = Local/[EMAIL PROTECTED] Cheers, Chad On Nov 15, 2004, at 7:59 AM, Shawn Dillon wrote: I have the AgentCallBackLogin working well when the support technician logs into the queue manually. If there a way to get certain extensions to automatically log into the queue? That way I do not have to worry about help desk staff forgetting to log into the support queue and never receiving support calls. As far as I know, it's enought to add those extensions as a members in queue.conf. -- Graf0 Ok, I will share more details with my particular installation. 1) In my extensions.conf I have the following; exten = 997,1,AgentCallBackLogin(999|[EMAIL PROTECTED]) 2) In my agents.conf I have the following group=1 agent = 999,1234,Test Agent 3) In my queues.conf I have he following member = Agent/999 When an agent dials extension 997 and enters their password they will then be included in the support queue for calls. If they do not call extension 997 and enter their password when a call is placed into the support queue a message appears on the console stating that no one is answering the support queue. Is there a way to get a certain extension to automatically log into a support queue? Or do I need to have every technician , at the start of every shift, log into the support queue manually? As an aside , this community has been very helpful in getting my Asterisk box up and running. Thanks to all. Shawn Dillon ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: OT: Grandstream problems
How do you downgrade the Budgetone to 10Mb? I don't see anything on the configuration page to do that. Also the specs on the Budgetone say it is a 10Base-T port. You can't. It already is 10MB. It can't do 100MB. Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help with this debug output?
Hi all, I've attached the output from asterisk with set verbose 3. During the time in the file, I placed two calls with my Zyxel 2000w to a Cisco 7912g. The first call worked fine, I was able to talk to the person on the other phone. The second call went through and rung the 7912g, but I was unable to hear the other person, and they could not hear me. This continues until I reset the 2000w, at which time, one call works again. Any ideas from this output as to why this is happening? -Chris Nov 15 12:33:01 DEBUG[6150]: chan_sip.c:5397 check_user_full: Setting NAT on RTP to 0 Nov 15 12:33:01 DEBUG[6150]: chan_sip.c:5401 check_user_full: Setting NAT on VRTP to 0 Nov 15 12:33:01 DEBUG[6150]: chan_sip.c:7227 handle_request: Check for res for wireless1 Nov 15 12:33:01 DEBUG[6150]: chan_sip.c:1671 update_user_counter: Call from user 'wireless1' is 1 out of 0 Nov 15 12:33:01 DEBUG[6150]: chan_sip.c:4612 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;transport=udp -- Executing Macro(SIP/wireless1-9409, stdexten|1000|SIP/cluster2) in new stack -- Executing Dial(SIP/wireless1-9409, SIP/cluster2|20) in new stack Nov 15 12:33:01 DEBUG[25615]: chan_sip.c:1297 create_addr: Setting NAT on RTP to 0 Nov 15 12:33:01 DEBUG[25615]: chan_sip.c:1301 create_addr: Setting NAT on VRTP to 0 Nov 15 12:33:01 DEBUG[25615]: chan_sip.c:1538 sip_call: Outgoing Call for cluster2 Nov 15 12:33:01 DEBUG[25615]: chan_sip.c:1671 update_user_counter: Call from user 'cluster2' is 1 out of 0 -- Called cluster2 Nov 15 12:33:01 DEBUG[6150]: chan_sip.c:860 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found Nov 15 12:33:01 DEBUG[6150]: chan_sip.c:860 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: Found -- SIP/cluster2-d469 is ringing Nov 15 12:33:02 DEBUG[6150]: chan_sip.c:810 __sip_ack: Acked pending invite 102 Nov 15 12:33:02 DEBUG[6150]: chan_sip.c:828 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Nov 15 12:33:02 DEBUG[6150]: chan_sip.c:4612 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;transport=udp -- SIP/cluster2-d469 answered SIP/wireless1-9409 -- Attempting native bridge of SIP/wireless1-9409 and SIP/cluster2-d469 Nov 15 12:33:02 DEBUG[25615]: rtp.c:1175 ast_rtp_write: Ooh, format changed from UNKN to ULAW Nov 15 12:33:03 DEBUG[6150]: chan_sip.c:828 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 1: Found Nov 15 12:33:03 DEBUG[25615]: rtp.c:1175 ast_rtp_write: Ooh, format changed from UNKN to ULAW Nov 15 12:33:08 DEBUG[25615]: rtp.c:190 send_dtmf: Sending dtmf: 51 (3), at 10.1.1.195 Nov 15 12:33:08 DEBUG[25615]: channel.c:1128 ast_settimeout: Scheduling timer at 160 sample intervals Nov 15 12:33:08 DEBUG[25615]: channel.c:1379 ast_read: Generator got voice, switching to phase locked mode Nov 15 12:33:08 DEBUG[25615]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Nov 15 12:33:09 DEBUG[25615]: channel.c:1388 ast_read: Auto-deactivating generator Nov 15 12:33:09 DEBUG[25615]: channel.c:1128 ast_settimeout: Scheduling timer at 0 sample intervals Nov 15 12:33:09 DEBUG[25615]: channel.c:2655 ast_channel_bridge: Didn't get a frame from channel: SIP/wireless1-9409 Nov 15 12:33:09 DEBUG[25615]: channel.c:2725 ast_channel_bridge: Bridge stops bridging channels SIP/wireless1-9409 and SIP/cluster2-d469 Nov 15 12:33:09 DEBUG[25615]: chan_sip.c:1767 sip_hangup: update_user_counter(cluster2) - decrement outUse counter Nov 15 12:33:09 DEBUG[25615]: app_dial.c:1029 dial_exec: Exiting with DIALSTATUS=ANSWER. == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/wireless1-9409' in macro 'stdexten' == Spawn extension (default-local-sip, 1003, 1) exited non-zero on 'SIP/wireless1-9409' Nov 15 12:33:09 DEBUG[25615]: chan_sip.c:1770 sip_hangup: update_user_counter(wireless1) - decrement inUse counter Nov 15 12:33:09 DEBUG[6150]: chan_sip.c:828 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 103: Found Nov 15 12:33:16 DEBUG[6150]: chan_sip.c:5397 check_user_full: Setting NAT on RTP to 0 Nov 15 12:33:16 DEBUG[6150]: chan_sip.c:5401 check_user_full: Setting NAT on VRTP to 0 Nov 15 12:33:16 DEBUG[6150]: chan_sip.c:7227 handle_request: Check for res for wireless1 Nov 15 12:33:16 DEBUG[6150]: chan_sip.c:1671 update_user_counter: Call from user 'wireless1' is 1 out of 0 Nov 15 12:33:16 DEBUG[6150]: chan_sip.c:4612 build_route: build_route: Contact hop: sip:[EMAIL PROTECTED]:5060;transport=udp -- Executing Macro(SIP/wireless1-d807, stdexten|1000|SIP/cluster2) in new stack -- Executing Dial(SIP/wireless1-d807, SIP/cluster2|20) in new stack Nov 15 12:33:16 DEBUG[26639]: chan_sip.c:1297 create_addr: Setting NAT on RTP to 0 Nov 15 12:33:16 DEBUG[26639]: chan_sip.c:1301 create_addr: Setting NAT on VRTP to 0 Nov 15
Re: [Asterisk-Users] T100P - Merlin Legend 100D not working
I still can't get asterisk and Merlin Legend to talk over the T1. If I dial a 44xx extension (should go to asterisk) I get no indication of a call in asterisk (even with pri debug on, the only clue is a POOL BUSY /OR OOS (Code 4C03) error in the Merlin's error log. Here is my config: Merlin Legend: -- A T1/PRI/BRI Clock Synchronization: A Primary Secondary Tertiary A 8/1 Loop ASlot # 0: CKE4 CPU ASlot # 1: 008 MLX ASlot # 2: 008 MLX ASlot # 3: 008 MLX ASlot # 4: 800 GS/LS ASlot # 5: 400 GS/LS/TTR ASlot # 6: 012 ASlot # 7: 012 ASlot # 8: 100D ASlot # 9: 408 GS/LS-MLX ASlot # 10: 408 GS/LS-MLX ASlot # 11: 012 ASlot # 12: 408 GS/LS-MLX ASlot # 13: 012 ASlot # 14: 100D-U ASlot # 15: 016 TRR Ringing Frequency - 20 Hz ASlot # 16: 008 MLX ASlot # 17: 100D A 873 17/ 1 No Remote 897 Yes Long 441 A 874 17/ 2 No Remote 897 Yes Long 441 A 875 17/ 3 No Remote 897 Yes Long 441 A 876 17/ 4 No Remote 897 Yes Long 441 A 877 17/ 5 No Remote 897 Yes Long 441 A 878 17/ 6 No Remote 897 Yes Long 441 A 879 17/ 7 No Remote 897 Yes Long 441 A 880 17/ 8 No Remote 897 Yes Long 441 A DS1 INFORMATION ADS1 SLOT ATTRIBUTES ASlot Type Format Supp Signal LineComp A 8 PRI ESF B8ZS DMI-MOS 1 A 14 PRI ESF B8ZS DMI-MOS 2 A 17 PRI ESF B8ZS DMI-MOS 1 A PRI INFORMATION A Slot 8 Switch: 4ESS (Long Distance ATT) A Slot 14 Switch: Legend-PBX (Switch 2) A Slot 17 Switch: Legend-Ntwk (Asterisk) A BchnlGrp #: Slot: TestTelNum: NtwkServ:Incoming Routing: A2 17 1 ElecTandNtwk Route Directly to UDP A Channel ID: 7 6 5 4 3 2 1 A LinePhoneNumber NumberToSend A873 A874 A875 A876 A877 A878 A879 A NON-LOCAL DIALPLAN A Range Ptn Dgt Range Ptn Dgt Range Ptn Dgt A1) 4112-4115 01 0418) 4182-4182 01 0435) - A2) 4118-4118 01 0419) 4190-4190 01 0436) - A3) 4121-4121 01 0420) 4192-4192 01 0437) - A4) 4127-4128 01 0421) 4194-4194 01 0438) - A5) 4131-4131 01 0422) 4196-4196 01 0439) - A6) 4135-4136 01 0423) 4198-4198 01 0440) - A7) 4138-4138 01 0424) 4200-4251 01 0441) - A8) 4143-4143 01 0425) 4252-4253 01 0442) - A9) 4147-4147 01 0426) 4254-4258 01 0443) - A 10) 4152-4153 01 0427) 4259-4260 01 0444) - A 11) 4156-4157 01 0428) 4261-4265 01 0445) - A 12) 4160-4160 01 0429) 4266-4280 01 0446) - A 13) 4162-4162 01 0430) 4281-4299 01 0447) - A 14) 4164-4164 01 0431) 4300-4358 02 0448) - A 15) 4165-4166 01 0432) 4360-4399 02 0449) - A 16) 4168-4174 01 0433) 4400-4410 03 0450) - A 17) 4177-4180 01 0434) - APattern 1: APool Absorb Other Digits FRL Call type A1)898- 00 3BOTH A2) -- -- A3) -- -- A4) -- -- APattern 2: APool Absorb Other Digits FRL Call type A1)894- 00 3BOTH A2) -- -- A3) -- -- A4) -- -- APattern 3: APool Absorb Other Digits FRL Call type A1)897- 00 3BOTH A2) -- -- A3) -- -- A4) -- -- - Asterisk: zaptel.conf: - span=1,1,0,esf,b8zs bchan=1-7 unused=8-23 dchan=24 fxsks=25-26 loadzone=us
Re: [Asterisk-Users] Cisco ATA and G729
On 16/11/2004 00:08 kido noagbodji said the following: i have an easy way to install the codec under FreeBSD? It was tough enough to install asterisk even with the FreeBSD ports. i do not believe that digium sells the g729 codecs for freebsd. however, i too am a freebsd user, and i guess what is needed is more people telling digium that we need the g729 codec on freebsd. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AU FreeBSD PRI Hardware
On 15/11/2004 22:46 Martin List-Petersen said the following: On Mon, 2004-11-15 at 07:46, Talbot Neil wrote: Hi, I was wondering if there is any PRI hardware that is Austel certified and works well with Asterisk under FreeBSD??? If anyone has any information please let me know as I seem to be having problems finding any documentation in regards to this. Isn't that the matter of finding a .au distributor/reseller, that did the certification ? i believe Australian Technology Partners in melbourne carry the digium cards. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA and G729
Dinesh Nair wrote: On 16/11/2004 00:08 kido noagbodji said the following: i have an easy way to install the codec under FreeBSD? It was tough enough to install asterisk even with the FreeBSD ports. i do not believe that digium sells the g729 codecs for freebsd. however, i too am a freebsd user, and i guess what is needed is more people telling digium that we need the g729 codec on freebsd. They do. It's considered unsupported and, oddly enough, that's the directory name it is located in (under the G729 directory) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 trunking - timing - ztdummy??
Håkan Källberg wrote: Hello! Perhaps someone can spread i little bit light on this: I want to trunk two Asterisk systems with each other. System A, behind a NAT-Firewall and System B with a real IP address. aix.conf on B: [mytrunk] host=dynamic username=mytrunk auth=md5 secret=yyy trunk=yes iax.conf on A: register = mytrunk:[EMAIL PROTECTED] When I make a reload an B I get the following: Nov 15 16:32:32 WARNING[-1244329040]: chan_iax2.c:6427 build_peer: Unable to support trunking on peer 'mytrunk' without zaptel timing I have downloaded the zaptel package, compiled it ( including ztdummy, which may be what I need ) and installed it. The kernel modules load: ztdummy 3492 0 zaptel228996 1 ztdummy crc_ccitt 2176 1 zaptel I don't know how to configure zaptel ( /etc/zaptel.conf ) to get this to work. I have no hardware, I only want timing for the IAX2 trunk ( and later on for Conference calls ). I have also read about the rtc package but have not tried it. I may have overseen very basic things... Please enlighten me! Regards: Håkan Try running zttest. Once zttest is working you should be OK. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with zyxel prestige 2002
This sounds odd. We use the same adapter. I will check this more.. Are u sure you have set the phone up correctly ? And also - have to checked the ring phone1 or phone2 on incomming calls ? / Stig Henning - Original Message - From: Mihkel Raba [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, November 14, 2004 9:51 PM Subject: [Asterisk-Users] problem with zyxel prestige 2002 Hi I tried to use Zyxel Prestige 2002 VoIP Analog Telephone Adapter with asterisk. Device registers both phones and i can call out. But incoming calls are not working. Asterisk - sip show peers shows zyxel, zyxel web interfce shows that devices are registered. But when i do incoming call to zyxel, phones do not ring and if voicemail is configured, calls go directly to voicemail. Any suggestions ? Mihkel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA and G729
Thanks Kido - Original Message - From: Eric Wieling [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 15, 2004 5:58 PM Subject: Re: [Asterisk-Users] Cisco ATA and G729 Dinesh Nair wrote: On 16/11/2004 00:08 kido noagbodji said the following: i have an easy way to install the codec under FreeBSD? It was tough enough to install asterisk even with the FreeBSD ports. i do not believe that digium sells the g729 codecs for freebsd. however, i too am a freebsd user, and i guess what is needed is more people telling digium that we need the g729 codec on freebsd. They do. It's considered unsupported and, oddly enough, that's the directory name it is located in (under the G729 directory) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MYSQL Dialplan Question
If you want good mysql/postgres/odbc/etc. support use http://svn.asteriskdocs.org/res_data/ Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shaun Tierney Sent: Monday, November 15, 2004 10:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] MYSQL Dialplan Question Well, I looked at the makefile and I could not see any options for SQL. So I did a grep for SQL on the distribution files and found that in version 0.7.0 support for MySQL was removed, so I'm guessing I'm just going to have to switch to Postgres or something. Thanks for the help, Shaun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Critchfield Sent: Monday, November 15, 2004 10:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MYSQL Dialplan Question On Mon, 2004-11-15 at 10:04 -0600, Shaun Tierney wrote: I am new to Asterisk, and I am having trouble connecting to the MySQL database located on the same machine as my Asterisk box. When the dialplan tried to connect to MySQL database, I get the following error message on the Asterisk console. Nov 15 09:29:41 WARNING[39760]: pbx.c:1279 pbx_extension_helper: No application 'MYSQL' for extension (default, s, 5) Here is the corresponding line in my dialplan. exten = s,5,MYSQL(Connect connid localhost admin abc123 asteriskdb) I am able to connect to the database with the same username and password using the MySQL console. Am I missing something in my installation or configuration that would cause this? Yes you are missing something in your installation. From reading the error message I can see you did not build the mysql application. My guess would be that it is not enabled in the makefile. Go edit it, compile, install, and try again. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MYSQL Dialplan Question
On 10:19 AM 11/15/2004, Michael Shuler wrote: If you want good mysql/postgres/odbc/etc. support use http://svn.asteriskdocs.org/res_data/ I may be incorrect, but I believe that res_data only lets you move configuration information into a database, however, what if you want to access databases and tables that have nothing at all to do with configuration data? The MYSQL application allows you to access any MySQL host.database.table. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Odd error at startup
Hi, I asked this before in the list amidst another question, and it got left behind. :-) Whenever I start asterisk with -gc, about 10 seconds passes and I get the following info: Nov 15 12:42:17 NOTICE[10369]: pbx_dundi.c:2841 destroy_trans: Peer '00:50:8b:f3:75:bb' has become UNREACHABLE! I assume this is innoculous, but I'm very curious what is causing this. The referenced Mac address doesn't exist in my environment (It's apparently a compaq NIC, and we don't have any). Gratz! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple TDM400 vs T1
Another question: We have a main office with approx 10 incoming lines. Some of the lines are now in a rotary configuration. Does anyone have any advice on the Pros/Cons to moving to a T1 pipe and an appropriate Digium card? Will the T1 give me more flexibility with Asterisk vs POTS? TIA Shawn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Avoiding 2 ring callerid delay for calls that don't go to voicemail
I have my dialplan configured for an incoming call on the FXO to connect right through to a FXS on the TDM100P. Because of the callerid the calling party gets 2 rings before asterisk picks up and then it's another 2 before the caller id shows up on the analog phone connected to the FXS module. I understand that the 2 rings are needed to collect CID, but is there any way to tell asterisk that since it's passing right through to the phone to streamline the process to 2 rings total instead of 4? People will almost be at the point of hanging up when I've just seen the CID at the phone for the first time. The only time the callerid would be used by Asterisk itself would be if the call goes to voicemail. Any hints? Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Odd error at startup
hi Whenever I start asterisk with -gc, about 10 seconds passes and I get the following info: Nov 15 12:42:17 NOTICE[10369]: pbx_dundi.c:2841 destroy_trans: Peer '00:50:8b:f3:75:bb' has become UNREACHABLE! this is sample entry for digium dundi node in dundi.conf. comment it out on dundi.conf and see www.dundi.com to learn what is dundi :) Matteo. -- Brancaleoni Matteo [EMAIL PROTECTED] Espia Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MC3810 IOS
I intend to use a MC3810 as a gateway to the pstn (ethernet -T1). I am curious if anyone has thoughts on what level of IOS will let me send calls from asterisk to the 3810 via sip? Thanks Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO setup
Hi, Thanks for your response. More info as requested: Location: USA FXO connection: Wipphone.com service (similar to Vonage) Analog Telephone Adaptor: Webphone WP200 FXO Card: X100P /etc/zaptel.conf fxsks=1 # X100P defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ks ; X100P echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=1 context=default ; Points to the default context of your extensions.conf channel = 1 On 15 Nov 2004 at 10:10, Rich Adamson wrote: Re: X100P Asterisk does not hangup automatically after caller leave a voicemail message and hangup.! Asterisk does not hangup automatically after the caller hangup in the Auto attendant menu system! What variables should I change to have * automatically hangup if the caller hangup? Right now, I have a variable set to a maximum of 60 seconds to hangup. All comments are greatly appreciated. Could be that you've not got the x100p configured correctly to detect the hangup; can't tell from what you've posted (twice). Assuming you actually get some sort of hangup notification from your telco, I'd look in zapata.conf samples for various ways to address it. Since we don't have a clue what country you're in, etc, no one is going to be able to help you. For voicemail, you might try maxsilence=10. Not a clue as to how to address your auto-attendant issue since you've not provide us with any configuration data whatsoever. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Odd error at startup
[EMAIL PROTECTED] wrote: hi Whenever I start asterisk with -gc, about 10 seconds passes and I get the following info: Nov 15 12:42:17 NOTICE[10369]: pbx_dundi.c:2841 destroy_trans: Peer '00:50:8b:f3:75:bb' has become UNREACHABLE! this is sample entry for digium dundi node in dundi.conf. comment it out on dundi.conf and see www.dundi.com to learn what is dundi :) Matteo. Thanks! (yes, I'm a newbie. Thanks for the gentle info). Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail shorter then (ex) 2sec - don't accept
On Mon, 2004-11-15 at 11:31 -0500, Seth Remington wrote: On Sun, 2004-11-14 at 13:02, Joseph wrote: In which configuration file I can specify that I don't want to accept messages for example shorter then 2sec. ? I've looked in voicemail.conf but I couldn't find any setting that will support this option. In most cases message shorter then 2 or 3sec will not contain any message and I don't want system to record them and sending an email to me. You were looking in the right config file. The parameter is called maxmessage. -Seth I just checked and I think this is not the one. maxmessage is to limit the message to the amount of time you specify in seconds. What I was looking for was to discard all the messages that are 3sec. or shorter. -- #Joseph ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple TDM400 vs T1
On Mon, 2004-11-15 at 11:54 -0700, Shawn Dillon wrote: Another question: We have a main office with approx 10 incoming lines. Some of the lines are now in a rotary configuration. Does anyone have any advice on the Pros/Cons to moving to a T1 pipe and an appropriate Digium card? Will the T1 give me more flexibility with Asterisk vs POTS? T1 will give you positive hangup and remote end answer information. At 10 lines, you should be close to the break even point between analog and T1 delivery. Also later upgrades in service like adding lines is just a matter of turning up the new pairs and reconfiguring asterisk. No new wiring. If you go the PRI route, it is possible to piggyback data on the unused lines until you need them for phone calls. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco ATA and G729
v 2004, at 18:58, Eric Wieling wrote: They do. It's considered unsupported and, oddly enough, that's the directory name it is located in (under the G729 directory) I have 2 g729 licenses runing into an PPC box with YDL very well against some other G729 machine codecs (cisco sipura) Adrià Vidal ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager API Call Origination Variables
Hi all, I am using the Asterisk Manager API to originate calls and it is working well, when a call is placed the local phone rings, once you pick it up you can here the call ringing the other end. Now, I am using Polycom IP 300 and I have them setup to auto-answer if I set the ALERT_INFO variable to Ring Answer. This works fine from my dial plan but I can't figure out how to set ALERT_INFO from the Manager API. Basically I want calls that are originated from the Manager API to automatically take place on the speaker phone. I have tried Action: SetVar Channel: sip/pete_desk Variable: ALERT_INFO Value: Ring Answer but it gives me about no such channel but this is the same channel I use to place the call immediately after attempting to set the variable. Any ideas? Thanks, Pete ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Voicemail shorter then (ex) 2sec - don't accept
On Mon, 2004-11-15 at 11:31 -0500, Seth Remington wrote: On Sun, 2004-11-14 at 13:02, Joseph wrote: In which configuration file I can specify that I don't want to accept messages for example shorter then 2sec. ? I've looked in voicemail.conf but I couldn't find any setting that will support this option. In most cases message shorter then 2 or 3sec will not contain any message and I don't want system to record them and sending an email to me. You were looking in the right config file. The parameter is called maxmessage. -Seth I just checked and I think this is not the one. maxmessage is to limit the message to the amount of time you specify in seconds. What I was looking for was to discard all the messages that are 3sec. or shorter. -- #Joseph Also a while back I noticed it did not understand that a message length equal to that of maxsilence was a null message and to discard it. David Cook ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO setup
Darly Coupet wrote: Hi, Thanks for your response. More info as requested: Location: USA FXO connection: Wipphone.com service (similar to Vonage) Analog Telephone Adaptor: Webphone WP200 FXO Card: X100P * */etc/zaptel.conf /fxsks=1 # X100P defaultzone=us loadzone=us / /etc/asterisk/zapata.conf /signalling=fxs_ks ; X100P echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=1 context=default ; Points to the default context of your extensions.conf channel = 1 / You are missing: busydetect=yes busycount=10 from your zapata.conf file. Just make sure they are above the channel = 1 line. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VICIDIAL in windows xp
Take a look at the astGUIclient FAQ: http://astguiclient.sourceforge.net/faq.html#2 you need to download the version of Time::HiRes that your perl version expects to be there(1.55 I think). You do not need a special ActivePerl version you can use any of the previous CPAN versions listed on this page with ActivePerl: http://search.cpan.org/~jhi/Time-HiRes-1.65/ MATT--- -Original Message- From: Guido Rebert [mailto:[EMAIL PROTECTED] Sent: Monday, November 15, 2004 12:22 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] VICIDIAL in windows xp hello everybody.. I´m trying to finallize with astguiclient/vicidial installation with the scratch-install . Everything works great but 2 stuffs I may see now. with astguiclient running on windows xp -- ok -VICIDIAL when I launch c:\AST_VICI\astVICIDIAL_0.8.pl an error appears and vicidial doesn´t launch.. at first perl asked for time::hires so I made ppm install Time::HiRes and installed HiRes version 1.49 So I tryied again but now askes for an newer version not available at active perl. this´s the error... time::hires object version 1.59 does not match $time::hires::XS_version 1.55 at c:/perl/lib/dynaloader.pm line 253 compilation failed in require at vicidial.pl line 83 begin failed--compilation aborted at vicidial.pl line 83 Someone knows how may I fix this? -ASTGUICLIENT - VICIDIAL web administration at welcome page (http://voip.local/astguiclient/welcome.php) there are a couplo of broken links... -http://voip.local/astguiclient/lists.php -http://voip.local/astguiclient/campaigns.php I didn´t found the pages.. anyone knows? Thanks to all... --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.788 / Virus Database: 533 - Release Date: 01/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Broadvoice number always busy
I am still getting a Busy message when I call in to my broadvoice number. Is anyone else still getting that or found a fix to it? I can call out all I want no problem. This seemed to start happening after the patch was applied. Jerry Geis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MC3810 IOS
Here is a sh version from my mc3810. I have it talking to asterisk via sip. Jojo --- MC3810-1#sh version Cisco Internetwork Operating System Software IOS (tm) MC3810 Software (MC3810-A2JK9SV5-M), Version 12.3(10), RELEASE SOFTWARE (fc3) Copyright (c) 1986-2004 by cisco Systems, Inc. Compiled Tue 17-Aug-04 07:35 by kellythw Image text-base: 0x00023000, data-base: 0x019C09BC ROM: System Bootstrap, Version 12.0(6r)T4, RELEASE SOFTWARE (fc1) ROM: MC3810 Software (MC3810-WBOOT-M), Version 12.0(1)XA4, EARLY DEPLOYMENT RELEASE SOFTWARE (fc1) MC3810-1 uptime is 2 weeks, 6 days, 13 hours, 9 minutes System returned to ROM by power-on System image file is flash:mc3810-a2jk9sv5-mz.123-10.bin From: [EMAIL PROTECTED] on behalf of Jason Brockman Sent: Mon 11/15/2004 11:03 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] MC3810 IOS I intend to use a MC3810 as a gateway to the pstn (ethernet - T1). I am curious if anyone has thoughts on what level of IOS will let me send calls from asterisk to the 3810 via sip? Thanks Jason winmail.dat___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail shorter then (ex) 2sec - don't accept
You were looking in the right config file. The parameter is called maxmessage. I just checked and I think this is not the one. maxmessage is to limit the message to the amount of time you specify in seconds. What I was looking for was to discard all the messages that are 3sec. or shorter. You are correct. I had it straight in my head but wrote the email wrong :) The parameter I originally meant was minmessage which should set the minimum length of the voicemail message in seconds. A quick source code check confirms that any voicemail less than minmessage will get deleted automatically. Sorry about the confusion. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] irq CPU state
Hi all, I have a particular install of a tdm400p (REV E/F) in which 'top' reports the irq CPU state is constantly 50%. ie: CPU states: cpuusernice systemirq softirq iowaitidle total0.7%0.0%0.5% 50.4% 0.0%0.0% 48.1% The above is _without_ Asterisk running. The heavy irq usage will cease the moment I rmmod wcfxs. The card has 2 fxo and 2 fxs ports. However, even if I comment out all the channels in zaptel.conf run ztcfg (ie: 0 channels configured), and then reinsert wcfxs ... the irq CPU state still jumps to 50%. I suspect that either the digium card is bad, or my motherboard is to blame (Asus A7N8X). Does this behaviour sound familiar to anyone? Thanks in advance, Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] $10 for G.729 ?
Hi, I have already send a message to Digium but I post this again to know of anyone who use the G.729 codec. My questions are: - Do I have to pay 10$ per month or it is only this fee to use the codec forever - How many licenses do I have to buy ? ( I have an Asterisk talking with two Cisco with VIC-2FXO each one, and one Cisco with a E1 ) - If I add another Cisco with another VIC-2FXO, Do I have to buy two more lincenses? - If I have 50 Cisco ATA Registered on my SIP.conf, Do I have to buy 50 licenses to use it when they talk between them without going through the PSTN? Thank you very much... Nahuel Ramos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Help with this debug output?
No one? On Mon, 15 Nov 2004 12:40:42 -0500, Chris TenHarmsel [EMAIL PROTECTED] wrote: Hi all, I've attached the output from asterisk with set verbose 3. During the time in the file, I placed two calls with my Zyxel 2000w to a Cisco 7912g. The first call worked fine, I was able to talk to the person on the other phone. The second call went through and rung the 7912g, but I was unable to hear the other person, and they could not hear me. This continues until I reset the 2000w, at which time, one call works again. Any ideas from this output as to why this is happening? -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Broadvoice number always busy
On Mon, 2004-11-15 at 15:01, Jerry Geis wrote: I am still getting a Busy message when I call in to my broadvoice number. Is anyone else still getting that or found a fix to it? I can call out all I want no problem. This seemed to start happening after the patch was applied. I've applied the patch on two separate * boxes (work and home) and both incoming and outgoing have been working fine. I'm using proxy.dca.broadvoice.com if that makes any difference to you. Does sip show registry show asterisk as registered with Broadvoice? -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] $10 for G.729 ?
The $10 is a permanent license. You only need a license for each *transcoding instance* in the asterisk box. The Cisco's already have 729 built in. Cisco to Cisco doesn't take any license, Cisco to voicemail will require 1 license per simultaneous voicemail connection. Cisco to conference room will require 1 license each, and so forth. tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nahuel Alejandro Ramos Sent: Monday, November 15, 2004 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] $10 for G.729 ? Hi, I have already send a message to Digium but I post this again to know of anyone who use the G.729 codec. My questions are: - Do I have to pay 10$ per month or it is only this fee to use the codec forever - How many licenses do I have to buy ? ( I have an Asterisk talking with two Cisco with VIC-2FXO each one, and one Cisco with a E1 ) - If I add another Cisco with another VIC-2FXO, Do I have to buy two more lincenses? - If I have 50 Cisco ATA Registered on my SIP.conf, Do I have to buy 50 licenses to use it when they talk between them without going through the PSTN? Thank you very much... Nahuel Ramos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and ISDN
Hello, Has anyone actually connect a BRI telephone to the BRI Card running Asterisk? I have been trying it and no luck so far. Here is my configuration: H/W: T1 Trunk, a few VOIP phones, a few analog phones and a few BRI Phones (Lucent i2021 phone and Tone Commander phone). S/W: Asterisk 1.0?, Linux 2.4-18, bri-stuff, wct4xxp and wcfxs S/W. Here is my problem: I connected the BRI phone to the BRI Card and the Q921 Layer was up. Then the phone send an Information Element (Q931 Msg) to the BRI Card. I think that IE message' data is [0x36 0x01 0x01] (it is Switch Hook Message). Then the Asterisk quicklly release the call (although I have not made a call connection). Anyway, I checked the q931.c file in libpri directory. In the q931_receive routine, the only Information Element message the routine handles is overlap dialling stuff. So, what ever that routine does not recongize, it sends release message right away. So, if anyone has ever connect the BRI phone to the BRI card with Asterisk, please let me know what S/W you are using. Thanks Ken On Mon, 2004-11-08 at 06:59, Gianni Veloce wrote: Dear * Experts, I intend to use a laptop for Asterisk at home (because of space problems and as I already have one spare). I would like Asterisk to 'sit between' my ISDN (BRI) Line and use my existing ISDN telephone as extension. After the hints from this list I learned that in order to do this I need a ISDN card capable of 'NT mode' for my telephone connection and another one (TE mode is enough) for connecting * to the BRI line. Needless to say that ISDN4Linux support is needed for both. That actually depends. There are two ways to archieve NT mode ISDN: chan_mISDN and bristuff (i've only dealt with bristuff). So no, you don't need ISDN4Linux support, but either mISDN or the zaphfc driver from bristuff. ISDN4Linux does not do NT mode, neither does chan_modem_i4l. chan_capi can't do NT mode either. Even tough chan_mISDN (and mISDN in the 2.6 kernel, which is a requirement for chan_mISDN) does support NT mode, it still needs the hardware to support it, too. The only cards supporting it are the ones based on Cologne Chip HFC-S chipset. Those are the same cards used for bristuff (which also utilises them on 2.4, no problems). In my case the question is: What ISDN cards (NT Mode capable) are existing in PCMCIA format (laptop) and where can I buy them? I haven't come across any PCMCIA HFC-S cards yet, but they exist. Kind regards, Martin List-Petersen -- ___ Find what you are looking for with the Lycos Yellow Pages http://r.lycos.com/r/yp_emailfooter/http://yellowpages.lycos.com/default.asp?SRC=lycos10 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk scalability IVR/Voicemail only
I have searched a bit on the Wiki and mailing list archives, but didnt see direct information regarding my scenario: 1. Asterisk for IVR/Voicemail ONLY (no PSTN, no MOH) 2. BudgeTone IP phones and HandyTone 286 ATAs 3. SIP only - separate Proxy+Registrar+CallRouter on other servers 4. G.711u codec, dtmfmode=rfc2883 5. No NAT/firewall (private ethernet network) What I'm looking for is scalability factors: 1. concurrent users accessing IVR and retrieving VM 2. concurrent mailboxes receiving VM (greeting playback, recording a msg) 3. impact of using configuration files vs. postgres vs. mysql Considering these specs for the Asterisk server, will adjust in accordance with scalability forecast: RHEL3 on single Xeon 3.2GHz, 4GB RAM Does anyone have general statistics/findings? Much appreciated! My apologies if this info is already out there somewhere in the archives. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Help with this debug output?
Other than the standard codec issues? No. disallow=all and allow=ulaw in [general] in sip.conf. NO other allow= lines. Chris TenHarmsel wrote: No one? On Mon, 15 Nov 2004 12:40:42 -0500, Chris TenHarmsel [EMAIL PROTECTED] wrote: Hi all, I've attached the output from asterisk with set verbose 3. During the time in the file, I placed two calls with my Zyxel 2000w to a Cisco 7912g. The first call worked fine, I was able to talk to the person on the other phone. The second call went through and rung the 7912g, but I was unable to hear the other person, and they could not hear me. This continues until I reset the 2000w, at which time, one call works again. Any ideas from this output as to why this is happening? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADSI questions for a 390 ADSI Phone
I got a 390 Adsi phone (unlocked) hooked to my wct400. It seems too work pretty well. I'm having two little problems with it. 1) The idle title screen will not show up unless I manually press service and select Asterix PBX. What seems odd is that if I do not manually select it it follows the script correctly if the phone goes off hook. Summing up, my title screen is the Time and date (when it should say Asterisk PBX and have a soft key for voice mail) and going off hook shows Asterisk PBX and 3 soft keys programmed. 2) Comedian mail always asks to download it self when it it already on the phone. In other news, is there documentation for the .adsi scripts anywhere? Thanks for all your help. --Rob ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Traffic shaping script for kernel 2.6 and SIP?
Hi! I want the SIP-traffic to have the highest priority. I guess the best method for this is traffic shaping. I'm using debian with kernel 2.6.5. I installed the tools tc and iptables but I'm not really sure how to use it. Can anybody help me in providing me a ready-made script? Thanks! Michael ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Manager API Call Origination Variables
On Mon, 15 Nov 2004, Peter Osborne wrote: I am using the Asterisk Manager API to originate calls and it is working well, when a call is placed the local phone rings, once you pick it up you can here the call ringing the other end. Now, I am using Polycom IP 300 and I have them setup to auto-answer if I set the ALERT_INFO variable to Ring Answer. This works fine from my dial plan but I can't figure out how to set ALERT_INFO from the Manager API. Basically I want calls that are originated from the Manager API to automatically take place on the speaker phone. I have tried Action: SetVar Channel: sip/pete_desk Variable: ALERT_INFO Value: Ring Answer The channel does not exist prior to the Originate action. However, you may be able to pass variables in the originate command itself: Action: Originate Channel: sip/12345 Exten: 1234 Context: default Variable: _ALERT_INFO=Ring Answer|SomeOtherVar=SomeOtherValue This may work. Peter ___ 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
