[Asterisk-Users] Loud echo with answer before dial
I'm having an echo related problem with Zap channels that are answered before a dial takes place, such as for IVR menus or fax detection. Basically, it sounds like the volume gets turned up to maximum while * is ringing the internal extensions waiting for someone to pick up, so when you pick up the phone to talk, you get an over-amplified echo. The sequence is like this: exten = s,1,Answer exten = s,2,Wait(2) exten = s,3,Dial(SIP/201SIP/202SIP/203SIP/204SIP/205SIP/206SIP/207) If you let it ring for a while before picking up, you get the horrible overdriven echo. It's like putting a really large value for rxgain, except that the volume goes down as the echo is trained out. Has anyone else noticed this? -- Seth cave quid dicis, quando, et cui Mattinen [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] Adding voice mail box
On Monday 19 July 2004 01:23 am, Brian K. West wrote: Dont have to.. just add it to the voicemail.conf and it will auto do everything for you. bkw Well, after having restarted * a few times, and rebooted once, I can say that no mailboxes were created automatically. I'm running a week old HEAD. Brian, what version were you running when you observed this nice feature? - Original Message - From: Steve [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 9:58 PM Subject: [Asterisk-Users] Adding voice mail box -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA+zjhljK16xgETzkRAh8jAKCJ7iJhFBVRxBFzbl8cGziqbnUjoQCdEzbb oTA7sXW1EXmmDGpUXrPf174= =zANK -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi: sending incoming calls to different contexts
Not sure if it works for you, but the simplest way is: [capi-in] exten = DIDNUM1,1,DoSomething exten = DIDNUM2,1,DoSomething exten = DIDNUM3,1,DoSomething where DIDNUMX is the direct indial number. Much nicer than goto statements with complicated splits. AFAIK you have only a DIDNUM if you have DID, that is with ISDN P2P, but not with P2MP. Or am I wrong? Are the multiple MSNs handled like DIDs? DID=Durchwahlnummern ___ Asterisk-Users mailing list [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] E100P and Colt Telecom (Europe)
Hi, Thanks a lot for the configs Fabe. I tried your zaptel.conf but I still get yellow and red alarms in zttool and * is unable to create any Zap channels (as expected with yellow and red alarms). I realise I will now have to start talking to Colt (in Ireland) to try and get the line up and running but if anyone has encountered this or something similar with Colt, or another provider in Europe, any tips would be greatly appreciated. Thanks, Aaron Message: 7 Date: Sat, 17 Jul 2004 10:38:26 +0200 From: Fabian Stelzer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] E100P and Colt Telecom (Europe) Reply-To: [EMAIL PROTECTED] zaptel.conf span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=nl zone=de doesen't work correctly for me :( but nl does... zapata.conf switchtype=euroisdn pridialplan=unknown signalling=pri_cpe group = 1 channel = 1-15,17-31 context=incoming this is the base config that works with colt... the rest has to be configured to you needs... Regards Fabe __ Do you Yahoo!? Vote for the stars of Yahoo!'s next ad campaign! http://advision.webevents.yahoo.com/yahoo/votelifeengine/ ___ Asterisk-Users mailing list [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] Brain-dead Grandstream BT102?
it never gets past the blue screen Ahhh, now I know: MICROSOFT is making the software for the Grandstream BT102. That explains something ... :-) ___ Asterisk-Users mailing list [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] E100P and Colt Telecom (Europe)
From the quote bits below: zaptel.conf span=1,0,0,ccs,hdb3,crc4 Assuming that it is the only E1 present, or the only one connected with the outside world, you should have the timing source configured: span=1,1,0,ccs,hdb3,crc4 Also, it might be that Colt are not using crc4 on your link, so try also with that removed: span=1,1,0,ccs,hdb3 Linus - Original Message - From: Aaron Clauson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 8:38 AM Subject: Re: [Asterisk-Users] E100P and Colt Telecom (Europe) Hi, Thanks a lot for the configs Fabe. I tried your zaptel.conf but I still get yellow and red alarms in zttool and * is unable to create any Zap channels (as expected with yellow and red alarms). I realise I will now have to start talking to Colt (in Ireland) to try and get the line up and running but if anyone has encountered this or something similar with Colt, or another provider in Europe, any tips would be greatly appreciated. Thanks, Aaron Message: 7 Date: Sat, 17 Jul 2004 10:38:26 +0200 From: Fabian Stelzer [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] E100P and Colt Telecom (Europe) Reply-To: [EMAIL PROTECTED] zaptel.conf span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 loadzone=nl zone=de doesen't work correctly for me :( but nl does... zapata.conf switchtype=euroisdn pridialplan=unknown signalling=pri_cpe group = 1 channel = 1-15,17-31 context=incoming this is the base config that works with colt... the rest has to be configured to you needs... Regards Fabe __ Do you Yahoo!? Vote for the stars of Yahoo!'s next ad campaign! http://advision.webevents.yahoo.com/yahoo/votelifeengine/ ___ Asterisk-Users mailing list [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] PSTN Gateway X101P
Marty Mastera [EMAIL PROTECTED] wrote: When I call the pstn number, the zaptel picks up the line on the first ring and then forwards it to the sip phone and rings it. Is there anyway to prevent the zaptel from picking up the line until the sip phone actully answers the call. This way I could answer the phone either locally on a regular analog handset or through the sip phone. The way it is now, it only rings my phones in the house 1 time. Hey Jason, glad things are working...I think I understand your problem and the short answer is no - there isn't a way to ring the x-lite without asterisk answering the call first (if I'm wrong about this, someone please correct me!). If you call Answer before Dial then Asterisk will answer the line before calling the device/softphone. If you don't call Answer then the line will not be picked up until the user of the device (or softphone) answers the call. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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] ast_data compile problem in asterisk CVS Asterisk CVS-HEAD-07/14/04
Hi all, Is there any updates on ast_data from svn.asteriskdocs.org/res_data to work with Asterisk cvs Asterisk CVS-HEAD-07/14/04? regards, Glynn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP client to IAXTel 800/888/877/866 dialing thru Asterisk
On Mon, 12 Jul 2004 11:42:57 -0700, Dameon D. Welch-Abernathy [EMAIL PROTECTED] wrote: Through my Asterisk server, I am trying to use IAXTel to dial 800-type numbers (yes, I know I can do the same thing with FWD and others via SIP, but I wanted to play with IAX a little). It appears I'm running into some sort of a codec mismatch or something because it's not working right. The SIP client is a SPA-3000. Phoneboy IAXcomm use gsm only that may help 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] Adding voice mail box
They only get created as they are used and voicemail left, try leaving a message and you should see that the structure etc is created. Steve -Original Message- From: Steve [mailto:[EMAIL PROTECTED] Sent: 19 July 2004 08:19 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Adding voice mail box On Monday 19 July 2004 01:23 am, Brian K. West wrote: Dont have to.. just add it to the voicemail.conf and it will auto do everything for you. bkw Well, after having restarted * a few times, and rebooted once, I can say that no mailboxes were created automatically. I'm running a week old HEAD. Brian, what version were you running when you observed this nice feature? - Original Message - From: Steve [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, July 18, 2004 9:58 PM Subject: [Asterisk-Users] Adding voice mail box -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA+zjhljK16xgETzkRAh8jAKCJ7iJhFBVRxBFzbl8cGziqbnUjoQCdEzbb oTA7sXW1EXmmDGpUXrPf174= =zANK -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ Asterisk-Users mailing list [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] Adding voice mail box
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 There's a script to create the mailbox with the asterisk source code: contrib/scripts/addmailbox On 19/07/2004, at 8:17 PM, Steve Hanselman wrote: They only get created as they are used and voicemail left, try leaving a message and you should see that the structure etc is created. - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA+6RqXeDVKqIr3GURAvVtAJ4s5Bc658HW6HDC/uehlhB95ZDyjwCbBHrw WlKE8cQzBrqzHY7XXVYEgj4= =BPGt -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LAN Switch w/ QoS
Does anyone have a recommendation for a 48 port LAN switch for a new * system? I'm not happy with NetGear's reliability. You can get Cisco 2950s for about $600/24 ports. And 48 ports from Dell for about the same price. I haven't used any of their latest round of switches, but their older ones were decent for the price. Cisco's switches are almost certainly better-made, but Dell's not *usually* that bad. I'd stay away from the Dell switches. Some models have serious issues that Dell says will never be fixed. (eg, they auto reboot if hit with some common html hijacking code, bad snmp request, etc, etc.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LAN Switch w/ QoS
Try picking up a 3com 3c17205 (4400 PWR) 24 port switch...it'll cost you a couple of bucks more but has POE built in and if you use the NJ200 (there's a guy selling boxes of 20 NJ200's on Ebay for $750 but he's got a lot so you can get him down to about $575) or NJ220 wall jacks (DONT GET THE NJ95,NJ100 or NJ105) it gives you tremendous flexibility, the NJ's are a proper managed 4 port switch in themselves and have built in QoS priority and the 4400 forwards POE to them and they forward it to the device). I managed to pick up some new 3c17205's for about $400 on Ebay (although they're scarce at the moment) and just finished wiring up my flat with the whole shebangmeans I've got phone points and network points everywhere in a single device and don't have to worry about power points. Have fun. P -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004, 4:22 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] LAN Switch w/ QoS Does anyone have a recommendation for a 48 port LAN switch for a new * system? I'm not happy with NetGear's reliability. You can get Cisco 2950s for about $600/24 ports. And 48 ports from Dell for about the same price. I haven't used any of their latest round of switches, but their older ones were decent for the price. Cisco's switches are almost certainly better-made, but Dell's not *usually* that bad. I'd stay away from the Dell switches. Some models have serious issues that Dell says will never be fixed. (eg, they auto reboot if hit with some common html hijacking code, bad snmp request, etc, etc.) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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] Brain-dead Grandstream BT102?
On Sun, 2004-07-18 at 23:52, Bruce Komito wrote: Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? When you eventually get the phone working - will you please share the knowledge with us on this forum? I'm also curious what you did to it to break it... power re-cycle whilst upgrading??? (I'd hate to do the same - as would others) -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [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] Polycom IP 500 Voicemail
Wiley E. Siler wrote: I have a solution that allows me to assign a soft key with no problems. However, it seems like a waste the the hard button labeled Voice Mail is not dialing right into voice mail. Is there a known way yo do this? I have tried everything in the manual but it doesn't seem to work. I have IP 500s and I want to be able to use all three display lines for just lines on the phone. I think that feature is inly available on the 1.2.0 sip firmware. It works on ours but when you press it, you still have to pick a line, then connect. The line button goes right to the voicemail. Also, do you know if it is possible to program the buttons along the bottom of the screen like normal soft buttons? Probably, but I haven't looked into it enough And finally... Is there a way to make the system dial without having to hit the Send key after dialing a number? look at the digitmap in sip.cfg -rb ___ Asterisk-Users mailing list [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 Sun/Monday News: Time to download, Scotty!
This week starts with the exciting news: We're getting close to Asterisk 1.0 again. After the failed attempt earlier this year, we've been able to remove a lot of the MAJOR/CRASH bugs from the bug tracker and Mark feel's it's time to target 1.0 again. At this point, the community needs to work as a community, spending extra time on finding bugs, solving issues, improving documentation and making Asterisk more stable. There has been extensive code reviews, but the more eyes that go through the source, the better. Telephony requires stability. Join the effort, regardless if you are a user, administrator, coder or documentation writer! Consider this: You're drafted! :-) This week's topics: --- * Asterisk 1.0: Second try * Astricon 2004: Early bird discount only applies in July * Changes in the #asterisk IRC channel - Registration required * Asterisk Developer of the week * Asterisk GUI of the week * Using Call parking with CVS head? Rename the config file! * Sunday News Re-run: Read the configs, Luke (even if you're an Asterisk guru) * Chan_sip2: Now updated * Recent CVS changes * Reporting bugs in the bug tracker *** Asterisk 1.0: Second try Last Saturday Mark Spencer released Asterisk 1.0 rc1. This is the first release candidate for Asterisk 1.0. A release candidate is software that needs extensive testing and a community that reports all bugs. I would like to send a *huge* THANK YOU to Mark from the Asterisk community. Mark has been working like a maniac to reach this point. The bug tracker has been rolling and rocking around with messages from Mark and patches have been produced and integrated at an incredible speed. During the weekend, the bug marshals decided to mark a lot of patches in the bug tracker post 1.0. Fixes will still be added, but we are trying to hold off all additional features for a while. We need to stabilize the 1.0 code, and one way to do that is to freeze the code and try to limit changes to bug fixes. There are a few unsolved major issues in the RC1. These are worked on and hopefully they will soon be resolved. The old stable CVS tree and distributions are no longer valid. The stable version is no longer supported. There's only one active CVS, the HEAD tree, and it's now the path forward to 1.0. Hopefully, we'll be able to branch into a stable and development CVS tree later on, after the 1.0 release. As always, this depends on if we can find a working solution for managing and maintaining the stable CVS tree. Most developers have a tendency of moving forward instead of maintaining stability and fixing bugs. However, a stable version is a much requested feature from many companies using Asterisk in a production environment. You can download Asterisk 1.0rc1 from several servers. There are also RPMs for the major Linux distributions. * Download mirrors: http://www.voip-info.org/wiki-Asterisk-mirrors * RPMs: ftp://ftp.nacs.net/asterisk To get better documentation for 1.0, join the asterisk-docs mailing list and contribute to the effort. Leif Madsen and Jared Smith really needs your help in order to get a decent handbook out to 1.0 release. * http://www.asteriskdocs.org *** Astricon 2004: Early bird discount only applies in July --- Astricon 2004 is getting closer. This is the first Asterisk user's and developer's conference. During July, you will get an early-bird discount on the registration fee so please do not forget to register early. Registering early also helps us planning. The more users, the bigger chance that a sponsor will decide to sponsor an Asterisk party :-) You may register for one, two or three days with or without hotel room booking at the web site. * http://www.astricon.net *** Changes in the #asterisk IRC channel - Registration required Due to abuse of the #asterisk IRC channel, the channel now requires that you are registred with FreeNode to be able to participate. Please /msg NickServ help register in your IRC client to learn how to register your nickname and get access to #asterisk. From the Freenode FAQ: Why should I register my nick? Your nick is how people on freenode know you. If you register it, you'll be able to use the same nick over and over. If you don't register, someone else may end up registering the nick you want. If you register and use the same nick, people will begin to know you by reputation. If they're running IRC software which supports CAPAB IDENTIFY-MSG, they'll be able to tell when someone is spoofing your identity. If a channel is set to mode +r, you won't be able to join it unless you are registered and identified to NickServ. If you try to join, you might be forwarded to a different channel. If a channel is set to mode +R, you won't be able to speak while on that channel unless you are registered and identified. Both of these modes are used to reduce channel harassment by DoS
[Asterisk-Users] TDM400P Internal Extenion Config
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new to
* so I have probably done something stupid!
I have a TDM400P with one FXO module and a FXS module. The main problem
I have is not being able to get the extension attached to the FXS module
to ring or be able to make calls. It gets a dialtone fine but I guess
this doesnt really mean all that much.
If I dial the extension I just get a 404 error on the phone
(Grandstream), but no errors at all on the console. I am using
CVS-HEAD-07/14/04. Here is a snippet of what I have in the various
config files.
zaptel.conf
loadzone=au
defaultzone=au
fxsks=1
fxoks=2
zapata.conf
[channels]
context=incoming
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
txgain=0.0
rxgain=0.0
group=1
callgroup=1
pickupgroup=1-4
immediate=no
busydetect=yes
busycount=7
callerid=asreceived
channel = 1
context=internal
group=2
signalling=fxo_ks
callerid=Fax 310
channel = 2
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
[sip]
exten = 301,1,Dial(SIP/Nick,20,tr)
exten = 302,1,Dial(SIP/Sharon,20,tr)
exten = 1000,1,Dial(SIP/NickSIP/Sharon,20,tr)
exten = 302,2,VoiceMail,u302
exten = 301,2,VoiceMail,u301
exten = 1000,2,VoiceMail,u
exten = 1000,102,VoiceMail,b
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain
include = outgoing
[incoming]
exten = s,1,Dial(SIP/NickSIP/Sharon,20,tr)
[outgoing]
exten = _7.,1,Dial(IAX2/login:[EMAIL PROTECTED]XXX/${EXTEN:1})
exten = 5.,1,Dial,Zap/1/${EXTEN:1}
[9103]
exten = 21060,1,Dial(SIP/Nick)
exten = 21062,1,Dial(SIP/Sharon)
[internal]
exten = 310,1,Dial,Zap/2
If I try to make any calls from the extension connected to the fxs
module i just get what sounds like a busy tone. Looking at the console
it generally give the error zt_set_hook: zt hook failed Device or
resource busy. It only gives this error when it goes off hook and
number dialed.
Only other information I can provide is a couple errors when asterisk
start up. I get the following:
Unable to open /dev/dsp: No such device
Unable to get our IP address, Skinny disable
Ignoring switchtype
Have not been able to dig out vast amounts of information on the above,
but what I have found didnt seem to point to my problem, but then what
do I know!
If anyone can help I would appreciate it! I'm going crazy here!
Kind regards
Nick
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[Asterisk-Users] Re: TDM400P Internal Extenion Config
Forgot to mention, both modules are show in ztcfg fine, see below:
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
Channel 02: FXO Kewlstart (Default) (Slaves: 02)
2 channels configured.
and zap show channel 2 give the following:
Channel: 2
File Descriptor: 19
Span: 1
Extension:
Dialing: no
Context: internal
Caller ID string: Fax 310
Destroy: 0
Signalling Type: FXO Kewlstart
Owner: None
Real: None
Callwait: None
Threeway: None
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Echo Cancellation: 128 taps, currently OFF
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Actual Hookstate: Onhook
Okay, so now I'm going to lie down in a dark room.
Cheers
Nick Cobley wrote:
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new
to * so I have probably done something stupid!
I have a TDM400P with one FXO module and a FXS module. The main
problem I have is not being able to get the extension attached to the
FXS module to ring or be able to make calls. It gets a dialtone fine
but I guess this doesnt really mean all that much.
If I dial the extension I just get a 404 error on the phone
(Grandstream), but no errors at all on the console. I am using
CVS-HEAD-07/14/04. Here is a snippet of what I have in the various
config files.
zaptel.conf
loadzone=au
defaultzone=au
fxsks=1
fxoks=2
zapata.conf
[channels]
context=incoming
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
txgain=0.0
rxgain=0.0
group=1
callgroup=1
pickupgroup=1-4
immediate=no
busydetect=yes
busycount=7
callerid=asreceived
channel = 1
context=internal
group=2
signalling=fxo_ks
callerid=Fax 310
channel = 2
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
[sip]
exten = 301,1,Dial(SIP/Nick,20,tr)
exten = 302,1,Dial(SIP/Sharon,20,tr)
exten = 1000,1,Dial(SIP/NickSIP/Sharon,20,tr)
exten = 302,2,VoiceMail,u302
exten = 301,2,VoiceMail,u301
exten = 1000,2,VoiceMail,u
exten = 1000,102,VoiceMail,b
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain
include = outgoing
[incoming]
exten = s,1,Dial(SIP/NickSIP/Sharon,20,tr)
[outgoing]
exten = _7.,1,Dial(IAX2/login:[EMAIL PROTECTED]XXX/${EXTEN:1})
exten = 5.,1,Dial,Zap/1/${EXTEN:1}
[9103]
exten = 21060,1,Dial(SIP/Nick)
exten = 21062,1,Dial(SIP/Sharon)
[internal]
exten = 310,1,Dial,Zap/2
If I try to make any calls from the extension connected to the fxs
module i just get what sounds like a busy tone. Looking at the console
it generally give the error zt_set_hook: zt hook failed Device or
resource busy. It only gives this error when it goes off hook and
number dialed.
Only other information I can provide is a couple errors when asterisk
start up. I get the following:
Unable to open /dev/dsp: No such device
Unable to get our IP address, Skinny disable
Ignoring switchtype
Have not been able to dig out vast amounts of information on the
above, but what I have found didnt seem to point to my problem, but
then what do I know!
If anyone can help I would appreciate it! I'm going crazy here!
Kind regards
Nick
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Re: [Asterisk-Users] Brain-dead Grandstream BT102?
same problem here. the display shows vulcan. - Original Message - From: Mark Elkins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 7:39 AM Subject: Re: [Asterisk-Users] Brain-dead Grandstream BT102? On Sun, 2004-07-18 at 23:52, Bruce Komito wrote: Following a(n apparently) failed attempt to upgrade a BT102, the phone is now brain-dead. Although it still has enough smarts to get a dhcp address and try to download the firmware and config, it never gets past the blue screen, nor will it respond to pings or port 80. Short of sending it back to Grandstream, is there any way to recover the phone? When you eventually get the phone working - will you please share the knowledge with us on this forum? I'm also curious what you did to it to break it... power re-cycle whilst upgrading??? (I'd hate to do the same - as would others) -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list [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] Numbering Plan and Siemens EWSD
Hi all, We're trying to hook up our Asterisk config (Card: TE410P) with a Siemens EWSD switch. The link is ok on both ends (green), with no errors. The problem is when we try to make a call from our side (via call files), we get the pri/E1 error Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) Our Telecom partner (they checked with Siemens) mentioned that we need to configure a dialplan as numbering plan (Rec. E.164) The stands for ISDN (Telephony), ISDN (Speech), etc This is what they told us, but the closest we can configure in Asterisk is the pridialplan (unknown, private, local, national, international). We tried all of them, with no difference. We also tried them with callerid set, no advance. Anyone familiar with this other dialplan, or with the integration of Asterisk/E1 with a Siemens EWSD switch. pri debug log of the call below (this was with pridialplan set to 'unknown') and without callerid. -- Making new call for cr 32780 Protocol Discriminator: Q.931 (8) len=32 Call Ref: len= 2 (reference 12/0xC) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'x' ] Sending Complete (len= 0) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32780/0x800C) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (Channel Identification) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32780/0x800C) (Terminator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) ] -- Processing IE 8 (Cause) -- Channel 1, span 1 got hangup Channel Zap/1-1 was never answered. NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 12/0xC) (Originator) Message type: RELEASE (77) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 32780/0x800C) (Terminator) Message type: RELEASE COMPLETE (90) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null ___ Asterisk-Users mailing list [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 Gui client
Hi, The version of astgui is 1.0.2. I am using PHP version 4.3.4-4 installed on a debian 3.0 system (testing) from apt-get. I do not have any GLOBAL_VARS set in my environment. What should it be? I am not very familiar with PHP. I had installed this on an existing system but made sure to install correctly all of the required packages that were listed in the instructions. I also have a problem, I dont know if it is related or not where when I first open the admin page I cannot get in with my username of gs102 and password of test. I verified that the username and password were in the database in the phones table. Thanks a lot! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: Friday, July 16, 2004 9:55 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Asterisk Gui client Hello, What version of the astguiclient suite are you using? What version of PHP are you using? Do you have GLOBAL_VARS turned on or off? It's very strange that being a POST all of the variables seem to be showing up on the URL like a GET would. also it doesn't sem to be submitting to the admin.php script like it should be. Did you follow the SCRATCH_INSTALL instructions or are you mostly installing this on an existing system? MATT--- PS- I wrote the astguiclient suite :) -Original Message- From: James Freire [mailto:[EMAIL PROTECTED] Sent: Friday, July 16, 2004 5:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Gui client I have installed the Asterisk gui client that is available off of sourceforge.net. I was curious if anybody here has used it and what experiences they have had with it. I am having a problem with it, I am able to use the admin page except when I try to submit information to the server to add phones I get an error, The requested URL /astguiclient/method=POST was not found on this server. The directory /astguiclient does exist and works because that is where the php files are located and running from. The URL for this command, so you can see what its submiting, is: http://172.16.200.80/astguiclient/method=POST?ADD=2extension=dialplan_numb er=voicemail_id=phone_ip=computer_ip=server_ip=login=pass=status=ACTI VEactive=Yphone_type=fullname=company=picture=submit=submit I am running Apache/1.3.29 with php installed also. My guess is that there is a bug somewhere in the php code but I do not know php well enough to troubleshoot it. Thanks a lot for any help, James Freire ___ Asterisk-Users mailing list [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] Adding voice mail box
On Mon, 2004-07-19 at 02:19, Steve wrote: On Monday 19 July 2004 01:23 am, Brian K. West wrote: Dont have to.. just add it to the voicemail.conf and it will auto do everything for you. bkw Well, after having restarted * a few times, and rebooted once, I can say that no mailboxes were created automatically. I'm running a week old HEAD. Brian, what version were you running when you observed this nice feature? Pretty much anything in the last year. Edit voicemail.conf, issue a reload and then LEAVE A VOICEMAIL. The mailbox won't actually be created until it needs to record a message. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unavailable/Withheld identification
Hi, I'm in the process of switching over to Asterisk from Alchemy kit and have hit a stumbling block. We're in the UK and use ISDN. At the moment we don't accept calls from withheld numbers (we just play them a message), but do accept calls from unavailable numbers. There doesn't seem to be any way for me to differentiate between the two number types in Asterisk (chan_CAPI) - they both appear to be presented as lacking callerID with no other identifier. I've had a look back through the archives and there doesn't seem to be an answer to this one. Does anybody have an idea on what to do or where to look? Many thanks, Nick. ___ Asterisk-Users mailing list [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] Unavailable/Withheld identification
Nick We are using QuadBRI cards from www.junghanns.net and also at home I have a couple of cheap HFC cards - in by Asterisk box sandwiched between my BT line and my Alchemy Cybergear Gold. Using this there is the possiblility to differentiate between Withheld and 'unavailable'. The latest code bri-stuff 0.1.0 supports it anyway. Never used Chan-capi but there is not really a reason to anyway, as the Junghanns drivers provide native Zaptel support. Best Regards Tim Robinson Tools Development Manager Motorola Ltd Midpoint Alencon Link BASINGSTOKE RG21 7PL United Kingdom Tel. +44 1256 790472 Fax+44 1256 790190 Mobile +44 7785 300316 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Barnes Sent: 19 July 2004 14:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Unavailable/Withheld identification Hi, I'm in the process of switching over to Asterisk from Alchemy kit and have hit a stumbling block. We're in the UK and use ISDN. At the moment we don't accept calls from withheld numbers (we just play them a message), but do accept calls from unavailable numbers. There doesn't seem to be any way for me to differentiate between the two number types in Asterisk (chan_CAPI) - they both appear to be presented as lacking callerID with no other identifier. I've had a look back through the archives and there doesn't seem to be an answer to this one. Does anybody have an idea on what to do or where to look? Many thanks, Nick. ___ Asterisk-Users mailing list [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] TE405P
Quoting hskim [EMAIL PROTECTED]: I have two questions. - Is TE410P is same as TE405P, or did I received different card? - zaptel.conf is configured CCS/HDB3. But It's configured as ESF/B8ZS. Hong TE4xx card are really cool they allow you to change type of interface either E1 or T1. By default cards are shipped in T1 mode all you need to do is change jumpers on the card to E1 and then the card will work as expected. m. ___ Asterisk-Users mailing list [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 w/ SIP response 481
OK, I think I have my problem narrowed down on my Avaya 4602SW SIP hardphone. When I reset the phone the phone works perfect up to the point until I get the following error in the CLI: -- Got SIP response 481 Call Does Not Exist back from my.home.external.ip This is how I have the SIP extension setup: [2002] type=friend username=2002 secret=mypassword host=dynamic context=from-sip mailbox=2002 nat=yes qualify=yes dtmfmode=info reinvite=no canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm callwaiting=1 The SIP phone is behind a wireless router with no ports forwarded to it, and the Asterisk server is straight on the Internet. I also have a Sipura behind the wireless router and it works just fine. If anyone could help me that would be great, this is driving nuts. Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI Dial, Extension dial SIP Loop
At the moment I'm prototyping an advanced ENUM application with PHP
fetched from LDAP. When a user enters a full hostname as SIP adress I get
loop problems from the AGI EXECUTE DIAL and from a Dial in the
extension.conf.
-- Executing AGI(SIP/1000-c3c3, enum.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/enum.php
enum.php: 123
enum.php: 3170327
enum.php: LDAP bind successful...
enum.php: telephoneNumber=3170327,ou=People,dc=eshara
enum.php: sip:[EMAIL PROTECTED]
enum.php: in: sip:[EMAIL PROTECTED]
enum.php: uit: sip/[EMAIL PROTECTED]
enum.php: in: sip:[EMAIL PROTECTED]
enum.php: uit: sip/[EMAIL PROTECTED]
-- AGI Script enum.php completed, returning 0
-- Executing Dial(SIP/1000-c3c3, sip/[EMAIL PROTECTED]) in new
stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 482 Loop Detected back from xxx.xxx.xxx.xxx
== No one is available to answer at this time
-- Executing Hangup(SIP/1000-c3c3, ) in new stack
== Spawn extension (default, 3170327, 3) exited non-zero on
'SIP/1000-c3c3'
But when I skip the @asterisk.blabla.bla it strangely works from the
extension.conf but not from the AGI script directly.
Now I set a variable, and then call do a:
AGI:
write(SET VARIABLE CALLTHIS .uri2tech($info[0]['description'][0]));
Extension:
Dial(${CALLTHIS})
-- AGI Script enum.php completed, returning 0
-- Executing Dial(SIP/1000-a5c0, sip/1000) in new stack
-- Called 1000
-- SIP/1000-d8a9 is ringing
== Spawn extension (default, 3170327, 2) exited non-zero on
'SIP/1000-a5c0'
I want to know why it fails with:
write(EXEC Dial .uri2tech($info[0]['description'][0]));
Is there a way to get this to work without stripping the hostname part?
How did other users solve this problem while using ENUM as backend and
calling locally?
Greetings,
Stefan de Konink
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[Asterisk-Users] FATAL: Module zaptel not found.
Dear Sirs, I'm running an Asterisk 0.9.1 in a Fedora Core 2 box. I installed a X100P card on my box and when i try to load modules i am rejected. When i type modprobe zaptel my Fedora respond : FATAL: Module zaptel not found. . The same uccurs when i type modprobe wcfxo May you help me. Thank you in advance Juanjo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help w/ SIP response 481
This is how I have the SIP extension setup: [2002] type=friend username=2002 secret=mypassword host=dynamic context=from-sip mailbox=2002 nat=yes qualify=yes dtmfmode=info reinvite=no canreinvite=no disallow=all allow=ulaw allow=alaw allow=gsm callwaiting=1 Not sure how to resolve your issue, but reinvite=no is not supported. canreinvite=no is the proper config option. Delete the reinvite. - B ___ Asterisk-Users mailing list [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] FATAL: Module zaptel not found.
Dear Sirs, I'm running an Asterisk 0.9.1 in a Fedora Core 2 box. I installed a X100P card on my box and when i try to load modules i am rejected. When i type modprobe zaptel my Fedora respond : FATAL: Module zaptel not found. . The same uccurs when i type modprobe wcfxo Perform an lsmod. See if it's listed. cat /proc/interrupts Also, by including relevent portions of the /etc/asterisk/zapata.conf and /etc/zaptel.conf, people would be in a better position to help you. - Brent ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF issue --help
On Fri, 2004-07-16 at 18:45, Andrew Yager wrote:
On 17/07/2004, at 3:24 AM, Eric Wieling wrote:
Tony Nichols wrote:
After calling a bank, or cc processing center; you have to enter
your
social security number, or the cc number - followed by the # key. The
lovely * voice responds transfering I'm sorry that was an invalade
selection. Sometimes the IVR on the other end still gets the digits
and
proceeds; other times it breaks the IVR on the bank side and hangs up.
How do I tell * to stop listning for the DTMF?
; dial a long distance outbound number exten =
_9XXX,1,Dial(${TRUNK}/${EXTEN:1},,Tt)
exten = _9XXX,2,Congestion
Stop telling it to listen to DTMF. It's pretty clear that you just
copied someone's Dial line from somewhere without learning what T and
t do. show application dial on the Asterisk CLI to learn what T and
t do.
If you really need the # transfer, there is a patch on the bug tracker
that implements the use of two keys for transfers (eg ##). I haven't
yet had a chance to test this feature, although I will.
Yours,
Andrew
Thank you Sir,
I'll give that a try I had thought the transfer option would have a
time limit ... not the duration of the call. The boss likes the
secretary to dial the number then transfer to his extension ... so I'll
try that patch -- or maybe make her put them on hold and tell the boss
what line to pickup.
t o n y
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RE: [Asterisk-Users] LAN Switch w/ QoS
Second that. Using stacked HP 2650 switches to support ~120 users and with QoS on, don't even have to worry about VLAN'ing the network. In HP/Compaq We Trust... I have been quite happy with our HP 2848 GigE switches that we put in for our desktops a few months ago. I have also used the 2650 48 10/100 + 2 GigE switches before. ___ Asterisk-Users mailing list [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] STILL NO AUDIO
I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved?
Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
On Fri, 2004-07-16 at 16:34, Christopher L. Wade wrote: Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also have voicemail and an extremely simple auto attendant setup to deal with calls during off hours. Due to the cost of all the components/software/consulting needed to make the NEC do everything it needs to do, we are hoping to 'merge' the NEC with an * box. In my 'working' * box, I have a wctdm11b (asterisk dev-kit) with 1 FXO and 1 FXS card. I say working, because I have everything setup if I totally bypass the NEC. As per an email conversation with Digium, we are connecting our POTS line to the FXS card, and the NEC to the FXO card. My current dilemma is that when I plug the * box and the NEC together, I cannot get the * box to 'dial' a particular extension on the NEC. It is my belief that this is due to some configuration changes needing to be made on the NEC. Unfortunately, this is the exact thing I needed to avoid and the reason for changing from the NEC to * in the first place. I know some changes to the NEC need to be made, but I am unsure as to exactly what, and how to do it. Any input on how to get this working would be greatly appreciated. If more information is required, please let me know. Please don't flame me for possibly being off-top, I don't think I need baby stepping through this, I simply need to know where to start looking. Thanks, Chris I know what ya mean I've spent nearly $800 in tech time for the Nec guy to help me get mine going. I have the Eletra 192 functioning right now, still have some bugs left but working. I used an Nec T1 card in the electra, and a digium t100p in my * box. Let me know if I can be of any help. When I get the last of the bugs worked out I plan to write down the details and put it on the wikki. t o n y ___ Asterisk-Users mailing list [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] Unavailable/Withheld identification
Speaking without experience of the exact combination you mention, but I'd expect that BT will send these to you using the combinations as follows: CLI: present Screening: available - Released number CLI: absent Screening: withheld - Withheld number CLI: absent Screening: available - Unavailable Number CLI: absent Screening: not available/interworking - Unavailable Number If you have access to the screening flag, this should help you. Linus - Original Message - From: Nick Barnes [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 2:07 PM Subject: [Asterisk-Users] Unavailable/Withheld identification Hi, I'm in the process of switching over to Asterisk from Alchemy kit and have hit a stumbling block. We're in the UK and use ISDN. At the moment we don't accept calls from withheld numbers (we just play them a message), but do accept calls from unavailable numbers. There doesn't seem to be any way for me to differentiate between the two number types in Asterisk (chan_CAPI) - they both appear to be presented as lacking callerID with no other identifier. I've had a look back through the archives and there doesn't seem to be an answer to this one. Does anybody have an idea on what to do or where to look? Many thanks, Nick. ___ Asterisk-Users mailing list [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] FATAL: Module zaptel not found.
On Mon, Jul 19, 2004 at 11:01:48AM -0300, PBX Portela wrote: Dear Sirs, I'm running an Asterisk 0.9.1 in a Fedora Core 2 box. I installed a X100P card on my box and when i try to load modules i am rejected. When i type modprobe zaptel my Fedora respond : FATAL: Module zaptel not found. . The same uccurs when i type modprobe wcfxo May you help me. Thank you in advance Juanjo It doesn't look like the module has been installed into the correct location in /lib/modules/kernel version/misc. You may need to build the zaptel driver. Good luck. O, en espan~ol: El modulo no esta' en un lugar correcto. Mire' en /lib/modules/su version del kernel/misc por zaptel.o o zaptel.ko. ?Necesitas compilar el 'driver' del zaptel? Buena suerte. -- Jayson Vantuyl ___ Asterisk-Users mailing list [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] TDM400P Internal Extenion Config
On Mon, 2004-07-19 at 07:13, Nick Cobley wrote: If I dial the extension I just get a 404 error on the phone (Grandstream), but no errors at all on the console. I am using CVS-HEAD-07/14/04. Here is a snippet of what I have in the various config files. Welcome to SIP. Dialtone is local to your phone and is not dependent on proper config. Hope that helps put you on the correct step to fix that problem. -- 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] STILL NO AUDIO
Happen to have any NAT in the mix? bkw -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian Nocetti Sent: Monday, July 19, 2004 9:25 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] STILL NO AUDIO I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved?
[Asterisk-Users] Cheap PoE switches/injectors?
I'm trying to spec out hardware for a new office, and I'd like to include power over Ethernet as an option. I've seen a handful of PoE injectors around $1000 for 24 ports and a couple switches up around $2500 for 24 ports. Are there any cheaper options, short of buying a boatload of 1-port injectors off of ebay? I don't really need more then 24 ports of PoE out of 48 total ports, so one of CIsco's big PoE switches is complete overkill. This is for a startup, where cheap is important. Thanks. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] STILL NO AUDIO
I suspect it will be solved when you put disallow=all and allow=ulaw in sip.conf and h323.conf (and NO OTHER ALLOW= LINES) On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved? -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Numbering Plan and Siemens EWSD
I'm using * connected to a EWSD too through an older zaptel (card T400P) and there is no problem with PRI calls. Did you put euroisdn? [EMAIL PROTECTED] wrote: Hi all, We're trying to hook up our Asterisk config (Card: TE410P) with a Siemens EWSD switch. The link is ok on both ends (green), with no errors. The problem is when we try to make a call from our side (via call files), we get the pri/E1 error Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) Our Telecom partner (they checked with Siemens) mentioned that we need to configure a dialplan as numbering plan (Rec. E.164) The stands for ISDN (Telephony), ISDN (Speech), etc This is what they told us, but the closest we can configure in Asterisk is the pridialplan (unknown, private, local, national, international). We tried all of them, with no difference. We also tried them with callerid set, no advance. Anyone familiar with this other dialplan, or with the integration of Asterisk/E1 with a Siemens EWSD switch. pri debug log of the call below (this was with pridialplan set to 'unknown') and without callerid. -- Making new call for cr 32780 Protocol Discriminator: Q.931 (8) len=32 Call Ref: len= 2 (reference 12/0xC) (Originator) Message type: SETUP (5) Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Unknown (67) '' ] Called Number (len=12) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) 'x' ] Sending Complete (len= 0) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 32780/0x800C) (Terminator) Message type: CALL PROCEEDING (2) Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (Channel Identification) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 32780/0x800C) (Terminator) Message type: DISCONNECT (69) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Temporary failure (41), class = Network Congestion (2) ] -- Processing IE 8 (Cause) -- Channel 1, span 1 got hangup Channel Zap/1-1 was never answered. NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 12/0xC) (Originator) Message type: RELEASE (77) Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 32780/0x800C) (Terminator) Message type: RELEASE COMPLETE (90) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null ___ Asterisk-Users mailing list [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] STILL NO AUDIO
No NAT, no FW, no nothing... from cisco 5300 with public ip without FW, to * with public ip without FW using SIP, and then from * to cisco 5300 without FW using chan_h323 De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de brianEnviado el: Lunes, 19 de Julio de 2004 11:39 a.m.Para: [EMAIL PROTECTED]Asunto: RE: [Asterisk-Users] STILL NO AUDIO Happen to have any NAT in the mix? bkw -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sebastian NocettiSent: Monday, July 19, 2004 9:25 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] STILL NO AUDIO I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved?
RE: [Asterisk-Users] STILL NO AUDIO
I WANT TO USE G729, I HAVE TO USE IT... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Eric Wieling Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO I suspect it will be solved when you put disallow=all and allow=ulaw in sip.conf and h323.conf (and NO OTHER ALLOW= LINES) On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved? -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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] CDR - Asterisk integration
I would be interested. Tenorio, Leandro wrote: Seshu, I'm interested could u provide more info... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu Sent: Wednesday, July 14, 2004 11:02 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] SMDR/CDR - Asterisk integration Hi All, The CDR Tool in .PHP is working great. We have put this into production. Here is the Link: http://67.109.153.236/asterisk-stat/cdr.php If anyone is interested, I will generously contribute the code for your use. Seshu Kanuri ___ Asterisk-Users mailing list [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: LAN Switch w/ QoS
For those on a low budget compex (http://cpx.com) has some very low cost switches that support QoS. http://www.cpx.com/proddetail.asp?c=Switchese=109 Bought a few of these myself, seem to work well. They are only manageable through an rs-232 console though, and don't have some other features of the high end switches like Spanning Tree protocol. Maron Kristofersson Colin Anderson wrote: Second that. Using stacked HP 2650 switches to support ~120 users and with QoS on, don't even have to worry about VLAN'ing the network. In HP/Compaq We Trust... I have been quite happy with our HP 2848 GigE switches that we put in for our desktops a few months ago. I have also used the 2650 48 10/100 + 2 GigE switches before. ___ Asterisk-Users mailing list [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: Updated Grandstream configurator
I was even considering going further and writing a crossplatform or a webapp for configuring. However I was thinking if someone has written some notes on the config file specification that could save a lot of time. I have no intention of competing with gsconfigure since I think it's an excellent app although I have to boot into windows to use it. Regards, Maron Holger Schurig wrote: Sorry about the delay getting back to you about the object creation error indicated in the snapshot. I am almost certain that the error is deriving from the creation of http objects by winhttp.dll. Microsoft's implementation of HTTP stuff for VB is really lame. Rewrite the app in Delphi. In Delphi, it's quite easy to write an app without Active-X, where all is compiled into the Program. HTTP access from inside Delphi is really no problem. This way you get one EXE file that you can deploy onto Win95, Win98, WinXT, Win2000, WinWhatever. And: older versions of Delphi are even free :-) ___ Asterisk-Users mailing list [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] BroadVoice problems?
Anyone else having problems with inbound Broadvoice this morning? -- Chris Tooley / Network and Development Services Networking Technologies Resource Center, LLC (NTRC) 8650 Spicewood Springs Road, Suite 105 Austin TX 78759 512-250-8985 / Fax 512-250-5909 www.ntrc.net / www.ntrcstore.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Inter-Tel Eclipse2 (IP PhonePlus)
Hello All, I am having the problem with Inter-Tel configuration... and really can't find source of it. I was able to configure the phone, it picks up the IP from our DHCP server and properly configured After that it registered to IPC card and I can make a call. but just for a few seconds. The phone keeps rebooting and reconfiguring IP address. What could be the problem? Vasyl Rublyov wrote: Can anyone tell me if this phone supports NAT ? If so then how? Ken Wiesner wrote: Vasyl, Not sure what kind of setup you're trying to do but if its a build out of an existing system you're two options are pretty much as follows: 1. Proprietary System Integration In this scenario you would use the Inter-Tel IPC. Supposedly they have a new card that supports SIP as well as their proprietary system, however I've been told it's buggy and quite expensive. This card plugs directly into the PBX just like a T1/Pri card or other hardware card. It has an Ethernet jack on it that you would connect directly to the Internet. It's mostly configurable via the web interface (its on some funky port, I think 8080 or 8008) however you will need to log into the DB Programming of the PBX to configure the circuit side of the PBX. We'll call this step "The Fun Part" as usually this results in some kind of massive breakdown of the PBX. If you go into DB Programming MAKE A BACKUP!!! I can't stress that enough. If the DB gets corrupt and you dont have a backup you will need to rebuild the entire PBX including dial plans, card configurations and voice applications. As for programming the IP keysets, you just hold down the 7 8 key while plugging in the Lan/Power cable in the back of the keyset. This is pretty straight forward, just follow the prompts on the keyset. Be sure to include 0's when entering IPs. i.e. 10.2.0.1 would be 010.002.000.001. Oh also, you will need the MAC addresses off all of the keysets as this is how the units are identified with the IPC. These get plugged into the IPC web interface under "Circuit Configurations." Pros: -If you need a solution that will give you the exact same functionality as if you were sitting at the office using the phone, this is probably your best solution. Cons: -The IP keysets are very picky about connectivity. If you don't have a solid high speed Internet connection they tend to be very choppy. If you beg the guys at corporate in Phoenix, they may send you the prequel software tool that lets you simulate their proprietary calls over the network. I would STRONGLY recommend testing with this before implementing this solution. -You can only network other IP Keysets and Softphones over the IPC. These will not support other hardware based IP phones. -The IPC and keysets are REALLY expensive. About $1,700 for the IPC (refurbished) and $700 per keyset!! (Ouch) 2. Asterisk Integration This other option which I think is a more powerful solution even though it lacks some features is getting a T1 card for the Inter-tel and an Asterisk Box with a Digium T100P. This will allow you to make a cross over cable and network the two. I'm not going to go into too many specifics here on how to do this because there is lots of documentation on the boards that tell you how to configure Asterisk. The only thing I can say is use Auto Line Build Out on the T1 of the Inter-tel and go for a high LBO on the T100P. The sample configurations say to use a low LBO but I've found it causes static and Red alarms. Pros: -You now have a fully customizable pbx module that you can use to create other apps such as IVRs and Unified Messaging applications. -Any SIP phone can now connect to the PBX via Asterisk server. -Low cost: Cost of computer plus $500 for T100P. Many keysets to choose from ranging in price. -LCR. Create a few channels on the T that will allow you to place calls from the PBX to a SIP carrier. (Can't do that with IPC) -Many other enhancements powered by Asterisk. Cons: -You're not going to have some of the functionality that you would have with the Inter-tel keysets. i.e. Agent management, Call monitoring, reverse transfers (call picking), paging and a few others. Anyways, we've tried both and 2 works out best for us. Hope this info helps! ~ken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Vasyl Rublyov Sent: Friday, July 02, 2004 7:23 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Inter-Tel Eclipse2 (IP PhonePlus) Thank you, Ken. I just asked because one from our clients is using this system. and I have to configure a few phones for connecting to their network/pbx. It really disaster to me. no SIP no docs... I just would like to cry a little :) and see if anyone can say anything good and informative about this system and that company :-) Thanks and have good weekends too. Ken Wiesner wrote: Vasyl, I'm reminded of the wedding scene in the movie "Old School"
Re: [Asterisk-Users] STILL NO AUDIO
Why don't you use asterisk-oh323? Michael. Sebastian Nocetti wrote: I WANT TO USE G729, I HAVE TO USE IT... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Eric Wieling Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO I suspect it will be solved when you put disallow=all and allow=ulaw in sip.conf and h323.conf (and NO OTHER ALLOW= LINES) On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved? ___ Asterisk-Users mailing list [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] TDM400P Internal Extenion Config
Thanks Steve, The SIP handsets are working find as I can make calls to other handsets as well as receive incoming calls via the FXO module. So all is good there. Cheers Nick Steven Critchfield wrote: On Mon, 2004-07-19 at 07:13, Nick Cobley wrote: If I dial the extension I just get a 404 error on the phone (Grandstream), but no errors at all on the console. I am using CVS-HEAD-07/14/04. Here is a snippet of what I have in the various config files. Welcome to SIP. Dialtone is local to your phone and is not dependent on proper config. Hope that helps put you on the correct step to fix that problem. ___ Asterisk-Users mailing list [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] ZyXEL 2000W
On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager [EMAIL PROTECTED] wrote: Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold, and the right button to take it off - except when I am in a call, these keys have no effect. I've tried teh 000c firmware, the 000e firmware and the Pulver 0011 firmware - but none work, so I'm wondering if this feature just simply isn't implemented, or if there is likely to be something wrong with my asterisk config. No it does not work, you need to use # transfer which will mean you will not be able to dial # into ivr's. Search on wiki for # transfer 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
[Asterisk-Users] Channel banks, voicemail, and immediate=no
When using a channel bank for analog handsets, you have a couple options in the way you handle transactions involving the analog handsets and origination. With immediate set to no, it appears to me that soon as a digit is pressed after going off-hook, the single digit is taken and processed against the context that the channel is associated with from the configuration in zapata.conf. With immediate set to yes, the extension s in the channel's context is processed. As far as I know, the method of handling channel bank based analog handsets is to use immediate=yes and then have extension s put the phone directly into a DISA command with no-password and a context for processing the entered calls. I have also tried in the past setting immediate=no, parsing off the first digit and sending the call into separate contexts (see example below) example with immediate=yes exten = s,1,DISA,no-password|internal example with immediate=no exten = 9,1,DISA,no-password|pstn-gateway In the first case, the problem I have is this: If I place the handset directly into DISA, how can I get stuttertone MWI indication? If I use the second method, in many cases, there is NO dialtone provided to the phone until after a dtmf entry is recieved. This I suspect is a channel bank issue because it seems to work on some banks, and not on others. Given the use of channel banks as a method to allow large number of analog phones to access an asterisk system, is there any way (or perhaps any interest in developing a method) to actually treat analog handsets on a channel bank like any other UA? In other words, why not have a method besides the two above so that I can stick the phones into a context (which understands it's for handling analog phones on a channel bank) that actually provides dial tone, and accepts dtmf until a match to the context extensions is found? In other words, with immediate=no, I'd like to see asterisk not jump on the first dtmf and try to match (going to i, if no match exists), but actually wait for as many dtmf's as required to match an extension in the context (e.g. exten = _1NXXNXX waits for 10 digits if dtmf 1 is the first digit). On a different track, am I doing something wrong above? For people who have configured channel banks for use with asterisk, have you found a 'perfect' configuration that you prefer to use? -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] STILL NO AUDIO
On Mon, 2004-07-19 at 10:00, Sebastian Nocetti wrote: I WANT TO USE G729, I HAVE TO USE IT... Not while testing you don't. Once you get it working with ULAW ONLY then see if you can get it working with G729. -- Useful Asterisk Docs (BOOKMARK THEM!): http://www.digium.com/index.php?menu=documentation (look at the Unofficial Links) and http://www.voip-info.org/wiki-Asterisk and http://www.fnords.org/~eric/asterisk/ (my site) and http://asteriskdocs.org/ ___ Asterisk-Users mailing list [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] STILL NO AUDIO
I WANT TO USE G729, I HAVE TO USE IT... When you have no FW and no NAT, then you seem to be inside your local network. In this case you shouldn't really care ?!?! ___ Asterisk-Users mailing list [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] Mac OS X installer for Asterisk
Benjamin, Is this package intended to mirror the directory structure of the linux builds? If so, I may have an issue: While /var/lib/asterisk is properly in place after running the installer, /usr/sbin/asterisk is not. I'm running on OS X 10.3.4 and downloaded the package on Sunday afternoon, if that's any help. Did I miss something? Thanks, Ted Wallingford -Original Message- From: Sunrise Ltd [mailto:[EMAIL PROTECTED] Sent: Saturday, July 17, 2004 2:09 PM To: astusr Subject: [Asterisk-Users] Mac OS X installer for Asterisk Hi I have created a Mac OS X installer package for installing Asterisk on OSX ver 10.2 and 10.3 Anyone who'd like to give this a try, please download the installer package from here ... http://www.astmasters.net/stuff/Asterisk.pkg.tgz to install Asterisk on OSX just double click the package file. please send any feedback to benjamin (at) sunrise (dash) tel (dot) com NOTE: this is a fairly old build but it's rock solid. We have run it on OSX Server 10.2.8 since October last year and it's been going like a Swiss clockwork. Rich Murphey has promised to fix the Makefile for the most recent CVS so it will build on OSX again. Once this is done, we'll make another installer package for the new version. Also, I am still working on extending the install package so that users can choose whether or not they want to install the sources. Anybody interested in this, please bare with me a few more days. regards benjamin -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __ Do You Yahoo!? http://bb.yahoo.co.jp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Wallingford, Ted.vcf Description: Binary data
Re: [Asterisk-Users] PhoneGaim?
On Sun, Jul 18, 2004 at 10:12:20AM -0500, Chris Howard wrote: I say on slashdot that the Linspire guys have released PhoneGaim. PhoneGaim is Gaim with SIP added on. Anyone want to add IAX2 as well... I'm writing a plugin for gaim right now that does iax2 on my off time. I haven't had much time to work on it lately, but I'm right now at kind of a decision point for what hooks will be in gaim to interface it. Maybe like a iaxtel/* protocol plugin. I'm still speculating about details though. I've got most of the lower stuff done now. Matthew Fredrickson ___ Asterisk-Users mailing list [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 problems?
Now that you mention it, yes... it seems that SIP isn't being passed from their PSTN gateway to the rest of their network... It's ringing, but there's no acknowledgement in * that anything's going on... - Original Message - From: Chris Tooley [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 8:19 AM Subject: [Asterisk-Users] BroadVoice problems? Anyone else having problems with inbound Broadvoice this morning? -- Chris Tooley / Network and Development Services Networking Technologies Resource Center, LLC (NTRC) 8650 Spicewood Springs Road, Suite 105 Austin TX 78759 512-250-8985 / Fax 512-250-5909 www.ntrc.net / www.ntrcstore.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] STILL NO AUDIO
Testing both... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Michael Manousos Enviado el: Lunes, 19 de Julio de 2004 12:25 p.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO Why don't you use asterisk-oh323? Michael. Sebastian Nocetti wrote: I WANT TO USE G729, I HAVE TO USE IT... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Eric Wieling Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO I suspect it will be solved when you put disallow=all and allow=ulaw in sip.conf and h323.conf (and NO OTHER ALLOW= LINES) On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved? ___ Asterisk-Users mailing list [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] TDM400P Internal Extenion Config
On Mon, 19 Jul 2004 20:13:09 +0800, Nick Cobley [EMAIL PROTECTED] wrote:
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new to
* so I have probably done something stupid!
Only a config issue I'm sure
[sip]
exten = 301,1,Dial(SIP/Nick,20,tr)
exten = 302,1,Dial(SIP/Sharon,20,tr)
exten = 1000,1,Dial(SIP/NickSIP/Sharon,20,tr)
exten = 302,2,VoiceMail,u302
exten = 301,2,VoiceMail,u301
exten = 1000,2,VoiceMail,u
exten = 1000,102,VoiceMail,b
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain
include = outgoing
add here
include = internal ; allow sip to dial 310
[incoming]
exten = s,1,Dial(SIP/NickSIP/Sharon,20,tr)
[outgoing]
exten = _7.,1,Dial(IAX2/login:[EMAIL PROTECTED]XXX/${EXTEN:1})
exten = 5.,1,Dial,Zap/1/${EXTEN:1}
[9103]
exten = 21060,1,Dial(SIP/Nick)
exten = 21062,1,Dial(SIP/Sharon)
[internal]
exten = 310,1,Dial,Zap/2
include = sip ; allow internal to dial sip phone
Try those changes and see how you get on
Jason
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[Asterisk-Users] IP Phone recommendation
Hi, I am looking for some affordable IP Phones. Any experiences with the SipToneII by ipDialog? What about soft phones? Any recommendations there (for Windoze and Linux)? Thanks, Yiannis ___ Asterisk-Users mailing list [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] Cheap PoE switches/injectors?
Look out for 3c17205 switches from 3com and read the QOS thread posting here at the moment. P -Original Message- From: Scott Laird [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004, 7:58 AM To: '[EMAIL PROTECTED]' [EMAIL PROTECTED] Subject: [Asterisk-Users] Cheap PoE switches/injectors? I'm trying to spec out hardware for a new office, and I'd like to include power over Ethernet as an option. I've seen a handful of PoE injectors around $1000 for 24 ports and a couple switches up around $2500 for 24 ports. Are there any cheaper options, short of buying a boatload of 1-port injectors off of ebay? I don't really need more then 24 ports of PoE out of 48 total ports, so one of CIsco's big PoE switches is complete overkill. This is for a startup, where cheap is important. Thanks. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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] STILL NO AUDIO
What kind of problem? All works OK except that config -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Holger Schurig Enviado el: Lunes, 19 de Julio de 2004 12:32 p.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO I WANT TO USE G729, I HAVE TO USE IT... When you have no FW and no NAT, then you seem to be inside your local network. In this case you shouldn't really care ?!?! ___ Asterisk-Users mailing list [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] Channel banks, voicemail, and immediate=no
I'm using a channel bank with a T1 card on the Asterisk server and have defined the FXS channels (user phones) to the context of [internal] and don't have any problems using the dial plans with the full digits. I haven't had any of them try to go to the i extension after the first digit. Not sure what configuration you're using that is causing this problem. I have immediate=no as well. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris A. Icide Sent: Monday, July 19, 2004 11:27 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Channel banks, voicemail, and immediate=no When using a channel bank for analog handsets, you have a couple options in the way you handle transactions involving the analog handsets and origination. With immediate set to no, it appears to me that soon as a digit is pressed after going off-hook, the single digit is taken and processed against the context that the channel is associated with from the configuration in zapata.conf. With immediate set to yes, the extension s in the channel's context is processed. As far as I know, the method of handling channel bank based analog handsets is to use immediate=yes and then have extension s put the phone directly into a DISA command with no-password and a context for processing the entered calls. I have also tried in the past setting immediate=no, parsing off the first digit and sending the call into separate contexts (see example below) example with immediate=yes exten = s,1,DISA,no-password|internal example with immediate=no exten = 9,1,DISA,no-password|pstn-gateway In the first case, the problem I have is this: If I place the handset directly into DISA, how can I get stuttertone MWI indication? If I use the second method, in many cases, there is NO dialtone provided to the phone until after a dtmf entry is recieved. This I suspect is a channel bank issue because it seems to work on some banks, and not on others. Given the use of channel banks as a method to allow large number of analog phones to access an asterisk system, is there any way (or perhaps any interest in developing a method) to actually treat analog handsets on a channel bank like any other UA? In other words, why not have a method besides the two above so that I can stick the phones into a context (which understands it's for handling analog phones on a channel bank) that actually provides dial tone, and accepts dtmf until a match to the context extensions is found? In other words, with immediate=no, I'd like to see asterisk not jump on the first dtmf and try to match (going to i, if no match exists), but actually wait for as many dtmf's as required to match an extension in the context (e.g. exten = _1NXXNXX waits for 10 digits if dtmf 1 is the first digit). On a different track, am I doing something wrong above? For people who have configured channel banks for use with asterisk, have you found a 'perfect' configuration that you prefer to use? -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 mailing list [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 problems?
I restarted asterisk and tried calling and it works now so either they fixed the problem and it's just a HUGE coincidence that I restarted * at the same time, or restarting * did the trick... P.S. What is the deal with the MailMan? When I send replies to the list, I've had it take up to 4 hours at most and on average it takes about an hour for anything to show up... -Chris - Original Message - From: Chris Tooley [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 8:19 AM Subject: [Asterisk-Users] BroadVoice problems? Anyone else having problems with inbound Broadvoice this morning? -- Chris Tooley / Network and Development Services Networking Technologies Resource Center, LLC (NTRC) 8650 Spicewood Springs Road, Suite 105 Austin TX 78759 512-250-8985 / Fax 512-250-5909 www.ntrc.net / www.ntrcstore.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Phone recommendation
On Mon, 2004-07-19 at 09:04, Yiannis Costopoulos wrote: Hi, I am looking for some affordable IP Phones. Any experiences with the SipToneII by ipDialog? So far our experience with the IP Dialog SipToneII is not good. It locks up after hang up on us, and just does not play nice. If anyone has any suggestions on how to get it working, we are all ears. The IP Dialog phone is running $200, while the Zip 4x4 is running $280-300 (depending on qty). We are deploying ~60 phones. Originally we were going to try and do 20 Uniden UIP200 and 40 Zip 4x4. We were unable to get our hands on a Uniden, and found that it would not even be available for an august deployment, so we decided to try the IP Dialog phone. The Uniden would have been a very worth while cost savings, as it's $150 and the Zip is $280 for our qty, but the $80 savings of the IP Dialog is not worth it to us. Harry What about soft phones? Any recommendations there (for Windoze and Linux)? Have not tried it but what about PhoneGaim? Thanks, Yiannis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Harry McGregor, Computing Manager Tucson Support Group - U.S. Geological Survey University of Arizona - Environment and Natural Resource Building 520-670-5574 (office) - [EMAIL PROTECTED] 520-661-7875 (Cell) - [EMAIL PROTECTED] The opinions/statements expressed herein are my own and should not be taken as a position, opinion, or endorsement of the University of Arizona or the U.S. Geological Survey. ___ Asterisk-Users mailing list [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 + NEC Electra Elite IPK Integration
Actually very straight forward. After calling Digium and getting some amazing tech support, I simply had to modify my dial string. The change was simply to make * pick up the line, and wait before dialing the requested extension. This was accomplished using several 'w' characters in the dial string before the extension. Once the remainder of the configuration of the * and the NEC is complete, and our NEC is set to pick up immediately, instead of waiting several rings, we should be able to use a 'clean' dial string, with no 'w' in it. Thanks, Chris Steve Totaro wrote: solution please. - Original Message - From: Christopher L. Wade [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 16, 2004 7:47 PM Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration Please disregard, I have 'solved' the issue. Thank you, Chris Christopher L. Wade wrote: Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also have voicemail and an extremely simple auto attendant setup to deal with calls during off hours. Due to the cost of all the components/software/consulting needed to make the NEC do everything it needs to do, we are hoping to 'merge' the NEC with an * box. In my 'working' * box, I have a wctdm11b (asterisk dev-kit) with 1 FXO and 1 FXS card. I say working, because I have everything setup if I totally bypass the NEC. As per an email conversation with Digium, we are connecting our POTS line to the FXS card, and the NEC to the FXO card. My current dilemma is that when I plug the * box and the NEC together, I cannot get the * box to 'dial' a particular extension on the NEC. It is my belief that this is due to some configuration changes needing to be made on the NEC. Unfortunately, this is the exact thing I needed to avoid and the reason for changing from the NEC to * in the first place. I know some changes to the NEC need to be made, but I am unsure as to exactly what, and how to do it. Any input on how to get this working would be greatly appreciated. If more information is required, please let me know. Please don't flame me for possibly being off-top, I don't think I need baby stepping through this, I simply need to know where to start looking. Thanks, 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 -- Christopher L. Wade Systems Administrator Unistar-Sparco Computers, Inc. dba Sparco.com 7089 Ryburn Drive Millington, TN 38053 USA Phone: (901) 872 2272 (800) 840 8400 Fax: (901) 872 8482 Email: [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 -- Christopher L. Wade Systems Administrator Unistar-Sparco Computers, Inc. dba Sparco.com 7089 Ryburn Drive Millington, TN 38053 USA Phone: (901) 872 2272 (800) 840 8400 Fax: (901) 872 8482 Email: [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] Cisco 7960 SIP V6 and distinctive ring.
Hi Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. Thanks in advance. P ___ Asterisk-Users mailing list [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] POE Switches and QOS
3com have some goods POE kit and some very nice managed wall jacks that supply POE and are fully managed. Here's an auction that the seller just closed:- http://cgi.ebay.co.uk/ws/eBayISAPI.dll?ViewItemcategory=40990item=5708757277rd=1ssPageName=WDVW Last time I spoke to him he had 5 boxes of 20 units and was willing to sell them at around $575 with a bit of arm twisting. The links on 3com are :- For the NJ200 (US Version) http://www.3com.com/products/en_US/detail.jsp?pathtype=purchasetab=featuressku=3CNJ200-CRM-20 and the NJ205 (Euro Version) http://www.3com.com/products/en_US/detail.jsp?tab=featurespathtype=purchasesku=3CNJ205-20 You can wall mount the Euro version so the ports are facing down so the unit looks very discrete. POE gets delivered on port 4 and as they are fully managed switches you can individually manage them and configure VLans (if you want to) allocating QOS priority to the port. The 3C17205 switch has full spanning tree and power management (i.e. you can kill the power on a specific port by logging onto the switch). As to loading for the switch you can read the specs here:- http://www.3com.com/products/en_US/detail.jsp?pathtype=purchasetab=featuressku=3C17205-US I may be telling people on here how to suck eggs but make sure whichever POE kit you get is 802.3af complient...a LOT of people are pushing products as POE but not using the proper standard (i.e. only using 5V or 12v up the wire which means voltage drop off is a problem over the cable length run). I've got no links to 3com just a happy customer...and POE stuff is starting to take off with 802.3af cameras and AP's starting to appear.makes managing a network a lot easier. P ___ Asterisk-Users mailing list [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 + NEC Electra Elite IPK Integration
I've actually just thought about using that solution. I realized over the weekend that my current solution has one VERY serious flaw in it. I forgot to mention that we, currently, have 24 phones/extensions in our office, we quite a few agents, many which are in multiple groups, we must be able to ring a fairly sizable portion of those extensions at any time. Using my current plan, a 'direct' line-for-line mapping from outside through my * box to the NEC, I won't be able to ring all necessary extensions if more than a few lines/channels are tied up between the * and the NEC. I think I'm going to end up using your solution of a NEC T1 card and one of the t100p cards so I can have more than 7 active extensions routing between the * box and the NEC. I would love it if you could go ahead and start putting your efforts online. I plan on doing the same once my configuration is done to our liking. Thanks, Chris Tony Nichols wrote: On Fri, 2004-07-16 at 16:34, Christopher L. Wade wrote: Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also have voicemail and an extremely simple auto attendant setup to deal with calls during off hours. Due to the cost of all the components/software/consulting needed to make the NEC do everything it needs to do, we are hoping to 'merge' the NEC with an * box. In my 'working' * box, I have a wctdm11b (asterisk dev-kit) with 1 FXO and 1 FXS card. I say working, because I have everything setup if I totally bypass the NEC. As per an email conversation with Digium, we are connecting our POTS line to the FXS card, and the NEC to the FXO card. My current dilemma is that when I plug the * box and the NEC together, I cannot get the * box to 'dial' a particular extension on the NEC. It is my belief that this is due to some configuration changes needing to be made on the NEC. Unfortunately, this is the exact thing I needed to avoid and the reason for changing from the NEC to * in the first place. I know some changes to the NEC need to be made, but I am unsure as to exactly what, and how to do it. Any input on how to get this working would be greatly appreciated. If more information is required, please let me know. Please don't flame me for possibly being off-top, I don't think I need baby stepping through this, I simply need to know where to start looking. Thanks, Chris I know what ya mean I've spent nearly $800 in tech time for the Nec guy to help me get mine going. I have the Eletra 192 functioning right now, still have some bugs left but working. I used an Nec T1 card in the electra, and a digium t100p in my * box. Let me know if I can be of any help. When I get the last of the bugs worked out I plan to write down the details and put it on the wikki. t o n y ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Christopher L. Wade Systems Administrator Unistar-Sparco Computers, Inc. dba Sparco.com 7089 Ryburn Drive Millington, TN 38053 USA Phone: (901) 872 2272 (800) 840 8400 Fax: (901) 872 8482 Email: [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] TDM400P Internal Extenion Config
Jason your a legend!!! I swear I tried include = internal in the sip
context, guess I managed to stuff it up somehow!!
Thanks so much for your help, sanity now saved :)
Regards
Nick
Jason Williams wrote:
On Mon, 19 Jul 2004 20:13:09 +0800, Nick Cobley [EMAIL PROTECTED] wrote:
Hopefully someone here can save my sanity. I have been trying to solve
this problem for days now, but just cant put my finger on it. Im new to
* so I have probably done something stupid!
Only a config issue I'm sure
[sip]
exten = 301,1,Dial(SIP/Nick,20,tr)
exten = 302,1,Dial(SIP/Sharon,20,tr)
exten = 1000,1,Dial(SIP/NickSIP/Sharon,20,tr)
exten = 302,2,VoiceMail,u302
exten = 301,2,VoiceMail,u301
exten = 1000,2,VoiceMail,u
exten = 1000,102,VoiceMail,b
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain
include = outgoing
add here
include = internal ; allow sip to dial 310
[incoming]
exten = s,1,Dial(SIP/NickSIP/Sharon,20,tr)
[outgoing]
exten = _7.,1,Dial(IAX2/login:[EMAIL PROTECTED]XXX/${EXTEN:1})
exten = 5.,1,Dial,Zap/1/${EXTEN:1}
[9103]
exten = 21060,1,Dial(SIP/Nick)
exten = 21062,1,Dial(SIP/Sharon)
[internal]
exten = 310,1,Dial,Zap/2
include = sip ; allow internal to dial sip phone
Try those changes and see how you get on
Jason
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[Asterisk-Users] X100P Call Waiting and Three Way Calling from SIP Device
I'm trying to be able to access the call waiting and three-way calls
features on a line attached to my X100P. For example, a party calls, the
X100P/Asterisk ring the 7960 on my desk, and all is fine. If I want to three
way call another individual in, I need to send a Flash to the X100P, and the
7960 doesn't appear to have any way to to that mid-call. All I can come up
with is transferring the call to a macro that will Flash, Dial the digits,
and return the call to me. For example, _*4. points to:
[app-flash]
exten = _*4.,1,Flash()
exten = _*4.,2,SendDTMF(${EXTEN:2})
exten = _*4.,3,Flash()
exten = _*4.,4,Transfer(1112)
This seems to work.. almost.. The flash, DTMF, and Flash commands work,
becuase the party on the first Zap call can hear the party on the second Zap
call. However, the Transfer back to the 7960 doesn't work, and after a few
seconds the entire call is dropped. Any idea on what I'm doing wrong? Is
there a simpler (in-call flash?) way to do this?
Ben Wern
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Re: [Asterisk-Users] Cheap PoE switches/injectors?
Hey man, I have a bunch of used power injectors that I would actually like to sell. If you are interested I will gather them all up and count them, but I know I have at least 24. I'd be glad to send you one to test. They are all Avaya/Lucent brand, they should work for any type of phone. Thanks, -Brian On Mon, 19 Jul 2004 09:03:49 -0700 (PDT), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Look out for 3c17205 switches from 3com and read the QOS thread posting here at the moment. P -Original Message- From: Scott Laird [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004, 7:58 AM To: '[EMAIL PROTECTED]' [EMAIL PROTECTED] Subject: [Asterisk-Users] Cheap PoE switches/injectors? I'm trying to spec out hardware for a new office, and I'd like to include power over Ethernet as an option. I've seen a handful of PoE injectors around $1000 for 24 ports and a couple switches up around $2500 for 24 ports. Are there any cheaper options, short of buying a boatload of 1-port injectors off of ebay? I don't really need more then 24 ports of PoE out of 48 total ports, so one of CIsco's big PoE switches is complete overkill. This is for a startup, where cheap is important. Thanks. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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] MAC OS X Panther :?
Just wondering if anyone has tried MAC OS X and panther. I will like to do SIP to H323, not sure if this will be possible on the MAC because of the Libraries PWlib and OPenh32 for Linux.. Just curious.. Anyway, anyone has an easy guide (step by step) to setup oh323 with asterisk. I saw a guide but i am not very savy on linux. thanks, Francisco - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, July 19, 2004 12:25 PM Subject: Asterisk-Users digest, Vol 1 #4598 - 14 msgs 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: STILL NO AUDIO (Michael Manousos) 2. Re: TDM400P Internal Extenion Config (Nick Cobley) 3. Re: ZyXEL 2000W (Jason Williams) 4. Channel banks, voicemail, and immediate=no (Chris A. Icide) 5. RE: STILL NO AUDIO (Eric Wieling) 6. Re: STILL NO AUDIO (Holger Schurig) 7. RE: Mac OS X installer for Asterisk (Wallingford, Ted) 8. Re: PhoneGaim? ([EMAIL PROTECTED]) 9. Re: BroadVoice problems? (Chris Shaw) 10. RE: STILL NO AUDIO (Sebastian Nocetti) 11. Re: TDM400P Internal Extenion Config (Jason Williams) 12. IP Phone recommendation (Yiannis Costopoulos) 13. Re: Cheap PoE switches/injectors? ([EMAIL PROTECTED]) 14. RE: STILL NO AUDIO (Sebastian Nocetti) --__--__-- Message: 1 Date: Mon, 19 Jul 2004 18:24:39 +0300 From: Michael Manousos [EMAIL PROTECTED] Organization: inAccess Networks To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] STILL NO AUDIO Reply-To: [EMAIL PROTECTED] Why don't you use asterisk-oh323? Michael. Sebastian Nocetti wrote: I WANT TO USE G729, I HAVE TO USE IT... -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Eric Wieling Enviado el: Lunes, 19 de Julio de 2004 11:46 a.m. Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] STILL NO AUDIO I suspect it will be solved when you put disallow=all and allow=ulaw in sip.conf and h323.conf (and NO OTHER ALLOW= LINES) On Mon, 2004-07-19 at 09:25, Sebastian Nocetti wrote: I cant do SIP - CHAN_H323 transmit audio!!! I can hear rings, but when connected, NOTHING It happened in both: SIP - CHAN_H323 and CHAN_H323 - SIP... when it will be solved? --__--__-- Message: 2 Date: Mon, 19 Jul 2004 23:26:06 +0800 From: Nick Cobley [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] TDM400P Internal Extenion Config Reply-To: [EMAIL PROTECTED] Thanks Steve, The SIP handsets are working find as I can make calls to other handsets as well as receive incoming calls via the FXO module. So all is good there. Cheers Nick Steven Critchfield wrote: On Mon, 2004-07-19 at 07:13, Nick Cobley wrote: If I dial the extension I just get a 404 error on the phone (Grandstream), but no errors at all on the console. I am using CVS-HEAD-07/14/04. Here is a snippet of what I have in the various config files. Welcome to SIP. Dialtone is local to your phone and is not dependent on proper config. Hope that helps put you on the correct step to fix that problem. --__--__-- Message: 3 Date: Mon, 19 Jul 2004 16:26:26 +0100 From: Jason Williams [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ZyXEL 2000W Reply-To: [EMAIL PROTECTED] On Fri, 16 Jul 2004 01:23:54 +1000, Andrew Yager [EMAIL PROTECTED] wrote: Does anyone have the call hold feature working? If you do... how did you make it work? The instructions say to press the left button to place the call on hold, and the right button to take it off - except when I am in a call, these keys have no effect. I've tried teh 000c firmware, the 000e firmware and the Pulver 0011 firmware - but none work, so I'm wondering if this feature just simply isn't implemented, or if there is likely to be something wrong with my asterisk config. No it does not work, you need to use # transfer which will mean you will not be able to dial # into ivr's. Search on wiki for # transfer Regards Jason --__--__-- Message: 4 Date: Mon, 19 Jul 2004 08:26:32 -0700 To: [EMAIL PROTECTED] From: Chris A. Icide [EMAIL PROTECTED] Subject: [Asterisk-Users] Channel banks, voicemail, and immediate=no Reply-To: [EMAIL PROTECTED] When using a channel bank for analog handsets, you have a couple options in the way you handle transactions involving the analog handsets and origination. With immediate set to no, it appears to me that soon as a digit is pressed after going off-hook, the single digit is taken and processed against
[Asterisk-Users] Flash Zap trunk from a Sipura
Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick up the hook-flash sent by the Sipura to answer that second call. I have configured the Sipura to send hook-flash messages to asterisk, and it does, but asterisk doesn't seem to know what to do with them. I've searched the wiki and google with no success. Thanks, Trevor ___ Asterisk-Users mailing list [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] Mac OS X installer: missing files fix
I've paraphrased the OS X installer developer's comments: there's a bug in Installer that is preventing the archive from working right. Below is the fix for the problem. First (obviously) run the installer. Since the executables are in the archive.pax.gz file in the installer package, first do a show package contents on the package file, then unstuff the enclosed archive.pax.gz file to the desktop... Then open up a shell, CD to the desktop, and run the following: cat Archive.pax | pax -r sudo cp -R usr/* /usr Anyway, hope this helps the Mac folks on the list. Thanks Benjamin for your efforts in this area. -Original Message- From: Wallingford, Ted [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 11:28 AM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Mac OS X installer for Asterisk Benjamin, Is this package intended to mirror the directory structure of the linux builds? If so, I may have an issue: While /var/lib/asterisk is properly in place after running the installer, /usr/sbin/asterisk is not. I'm running on OS X 10.3.4 and downloaded the package on Sunday afternoon, if that's any help. Did I miss something? Thanks, Ted Wallingford -Original Message- From: Sunrise Ltd [mailto:[EMAIL PROTECTED] Sent: Saturday, July 17, 2004 2:09 PM To: astusr Subject: [Asterisk-Users] Mac OS X installer for Asterisk Hi I have created a Mac OS X installer package for installing Asterisk on OSX ver 10.2 and 10.3 Anyone who'd like to give this a try, please download the installer package from here ... http://www.astmasters.net/stuff/Asterisk.pkg.tgz to install Asterisk on OSX just double click the package file. please send any feedback to benjamin (at) sunrise (dash) tel (dot) com NOTE: this is a fairly old build but it's rock solid. We have run it on OSX Server 10.2.8 since October last year and it's been going like a Swiss clockwork. Rich Murphey has promised to fix the Makefile for the most recent CVS so it will build on OSX again. Once this is done, we'll make another installer package for the new version. Also, I am still working on extending the install package so that users can choose whether or not they want to install the sources. Anybody interested in this, please bare with me a few more days. regards benjamin -- Sunrise Telephone Systems Ltd 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan __ Do You Yahoo!? http://bb.yahoo.co.jp/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Wallingford, Ted.vcf Description: Binary data
Re: [Asterisk-Users] Cheap PoE switches/injectors?
On Jul 19, 2004, at 9:03 AM, [EMAIL PROTECTED] wrote: Look out for 3c17205 switches from 3com and read the QOS thread posting here at the moment. So $1600 for 24 ports. That's not *too* bad. HP seems to have a similar model (2626-PWR) for a similar price. 3com also seems to have a 24-port injector for $800. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Gui client
Hi All, Please checkout the following GUI web panels, which have been created and installed from the source code available in this forum. http://67.109.153.236/*web/ It edits extensions.conf after some customization.However unable to update sip.conf. http://67.109.153.236/asterisk-stat/cdr.php Link to the CDR Tool. http://67.109.153.236/cgi-bin/am/am-main.pl The perl based Asterisk GUI Management system. Help is available online in same panel. This code is a bit cumbersome and I am not going to attempt developing this. PHP is much more preferrable. http://67.109.153.236/cgi-bin/astcc/astcc-admin.cgi Calling card application is installed. Uses database `asteriskcc`. Unable to get make it run though, to check it's technical functionality. Once the code reaches some useful level, I am going to post the source code back here, through a download link. Seshu Kanuri -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Freire Sent: Friday, July 16, 2004 5:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Gui client I have installed the Asterisk gui client that is available off of sourceforge.net. I was curious if anybody here has used it and what experiences they have had with it. I am having a problem with it, I am able to use the admin page except when I try to submit information to the server to add phones I get an error, The requested URL /astguiclient/method=POST was not found on this server. The directory /astguiclient does exist and works because that is where the php files are located and running from. The URL for this command, so you can see what its submiting, is: http://172.16.200.80/astguiclient/method=POST?ADD=2extension=dialplan_number=voicemail_id=phone_ip=computer_ip=server_ip=login=pass=status=ACTIVEactive=Yphone_type=fullname=company=picture=submit=submit I am running Apache/1.3.29 with php installed also. My guess is that there is a bug somewhere in the php code but I do not know php well enough to troubleshoot it. Thanks a lot for any help, James Freire ___ Asterisk-Users mailing list [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 + NEC Electra Elite IPK Integration
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20, I don't think I'll need the U30, but I'm not entirely sure. Thanks, Chris Tony Nichols wrote: On Fri, 2004-07-16 at 16:34, Christopher L. Wade wrote: Hi, I'm am currently in the process of trying to integrate an * box with an NEC Electra Elite IPK. Currently, we have 7 POTS lines coming into our building. These lines are plugged into our NEC using the appropriate analog line interface card from NEC. The NEC effectively has NO configuration done to it, other than to make all the internal phones ring when a call comes in. We also have voicemail and an extremely simple auto attendant setup to deal with calls during off hours. Due to the cost of all the components/software/consulting needed to make the NEC do everything it needs to do, we are hoping to 'merge' the NEC with an * box. In my 'working' * box, I have a wctdm11b (asterisk dev-kit) with 1 FXO and 1 FXS card. I say working, because I have everything setup if I totally bypass the NEC. As per an email conversation with Digium, we are connecting our POTS line to the FXS card, and the NEC to the FXO card. My current dilemma is that when I plug the * box and the NEC together, I cannot get the * box to 'dial' a particular extension on the NEC. It is my belief that this is due to some configuration changes needing to be made on the NEC. Unfortunately, this is the exact thing I needed to avoid and the reason for changing from the NEC to * in the first place. I know some changes to the NEC need to be made, but I am unsure as to exactly what, and how to do it. Any input on how to get this working would be greatly appreciated. If more information is required, please let me know. Please don't flame me for possibly being off-top, I don't think I need baby stepping through this, I simply need to know where to start looking. Thanks, Chris I know what ya mean I've spent nearly $800 in tech time for the Nec guy to help me get mine going. I have the Eletra 192 functioning right now, still have some bugs left but working. I used an Nec T1 card in the electra, and a digium t100p in my * box. Let me know if I can be of any help. When I get the last of the bugs worked out I plan to write down the details and put it on the wikki. t o n y ___ Asterisk-Users mailing list [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] Codecs - Advantages
Hi, I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [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] Polycom IP 500 Voicemail
Thank you! Can you tell me more about the dial plan feature? How do you setup the correct digitmap? W -Original Message- From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 4:56 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail Wiley E. Siler wrote: I have a solution that allows me to assign a soft key with no problems. However, it seems like a waste the the hard button labeled Voice Mail is not dialing right into voice mail. Is there a known way yo do this? I have tried everything in the manual but it doesn't seem to work. I have IP 500s and I want to be able to use all three display lines for just lines on the phone. I think that feature is inly available on the 1.2.0 sip firmware. It works on ours but when you press it, you still have to pick a line, then connect. The line button goes right to the voicemail. Also, do you know if it is possible to program the buttons along the bottom of the screen like normal soft buttons? Probably, but I haven't looked into it enough And finally... Is there a way to make the system dial without having to hit the Send key after dialing a number? look at the digitmap in sip.cfg -rb ___ Asterisk-Users mailing list [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 7960 SIP V6 and distinctive ring.
On Jul 19, 2004, at 9:29 AM, [EMAIL PROTECTED] wrote:
Hi
Can anyone with distinctive ring on their 7960's possibly post how
they've got it to work?
I understand that the ALERT_INFO variable is involved but using the
examples for the variable value from the WiKi I'm just getting an
error message from the Asterisk concole.
I'm setting it to 'Bellcore-dr1' through 'Bellcore-dr4'. I'm grabbing
the value out of Asterisk's database and sticking it into ALERT_INFO
like this:
[macro-setalertinfo]
exten = s,1,DBGet(ALERT_INFO=distinctivering/${CALLERIDNUM})
Works fine for me. You should also be able to do
'SetVar(ALERT_INFO=Bellcore-dr1)' without problems. Can you show us
the line that's generating errors?
Scott
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[Asterisk-Users] CTR21/CTR37 Gigaset phones and GS HT286
I'm having no end of trouble with some Siemens Gigaset phones and GS HT286s. Gigaset 100 and 3010 phones work perfectly, but a 4010 only rings once then it goes off and then just flashes it's LEDs and displays incoming call on the LCD with no further ringing. According to the manual it is CTR37 but the only setting on the GSs is CTR21, I've tried different cables but some actually make it worse i.e. no ring at all. -- Dave Cotton [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] Cheap PoE switches/injectors?
Has anyone tried the new dlink powered switches? I remember seeing an online voip store selling these as a good option for providing power in a voip application. They were price at 1100 for a 24 port model. The lowest cost solution I have seen are the individual 3com power injectors which can be had for between $16-$25. I have done some minimal testing with one for use with wireless access points and it seems workable, although not a good solution for a high density environment. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Scott Laird [EMAIL PROTECTED]: On Jul 19, 2004, at 9:03 AM, [EMAIL PROTECTED] wrote: Look out for 3c17205 switches from 3com and read the QOS thread posting here at the moment. So $1600 for 24 ports. That's not *too* bad. HP seems to have a similar model (2626-PWR) for a similar price. 3com also seems to have a 24-port injector for $800. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ Asterisk-Users mailing list [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 gateway implementation?
Hello, I need help in creating a simple PSTN Gateway. This is the scenario: -I have one client sending me VoIP traffic (they dont have asterisk, so IAX is out of the picture for me) and I need to validate that traffic (only accept calls coming from his IP). After that I would terminate the calls to the PSTN network and keep logs for billing purposes. -I have a TE405P board and only one T1 worth of phone lines (24) connected to it using an Adtran TA750 channel bank. Is Asterisk capable of handling multiple incoming VoIP calls arriving from the same source (IP) or do I need to get something else to take the incoming traffic and pass it on to Asterisk? (Ive read about using SER as a SIP proxy, but its not clear to me wheather I need it or not). Can I use the OpenH.323 module to take care of the incoming VoIP traffic? I am totally new to all this. Any help will be really appreciated. Please give me your input on which is the best implementation of a VoIP-PSTN gateway and the some sample configuration files for the plattforms involved (Asterisk, SER, OpenH323, etc.) Thanks in advance. Alejandro Sosa.
Re: [Asterisk-Users] Cheap PoE switches/injectors?
Scott Laird wrote: So $1600 for 24 ports. That's not *too* bad. HP seems to have a similar model (2626-PWR) for a similar price. 3com also seems to have a 24-port injector for $800. I still don't understand why I can buy single-port injectors for $20, but multi-port models are $30 per port and up. You'd think that having a single combined power supply and other bits would reduce the cost, not increase it. I'd like to see someone make a single-port injector that fits into a keystone jack, so I can insert 24 of them into a 24-port keystone rack-mount panel, and then power them with a stock Valcom 48V A-battery power supply. ___ Asterisk-Users mailing list [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] Polycom IP 500 Voicemail
Thank you! Can you tell me more about the dial plan feature? How do you setup the correct digitmap? Check the Administrator's Document. You can find it on the Wiki, under IP Phones.. Polycom. Did you try to look up the digitmap feature before sending this post? If not, you should be able to understand it when you read it, it's relatively straight forward. No one can setup a correct digitmap for you, as it will vary greatly on how you have setup your PBX. - Brent ___ Asterisk-Users mailing list [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] Codecs - Advantages
If you have the bandwidth then use ulaw :) bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, July 19, 2004 12:44 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Codecs - Advantages Hi, I'm planning to use a Asterisk with Digium E1 cards, I understand that using a codec such as G.729 can be very CPU demanding. What are the real advantages of using a codec such as G.729 ? Bandwidth only ? Using no compression wouldn't increase the scalability of my asterisk PBX ? This is considering I have no bandwidth issues in my network. Thanks __ Todavía no tenés tu Ciudad Internet Mail? Obtenelo ahora! - http://webmail.ciudad.com.ar Descargá Gratis el nuevo Internet Explorer 6.0, el mejor software para actualizar tu PC. http://www.ciudad.com.ar/ar/servicios/ie/ ___ Asterisk-Users mailing list [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] Polycom IP 500 Voicemail
Strange, I have an IP500 right out of the new-plastic-gadget-smell box, and it doesn't have a button labelled Voicemail. On the left side are the blue speaker, red mute, and blue headset buttons, then next to them top to bottom are the three Line buttons (clear covers for putting your own labels), Directories, Services, Call Lists, Conference, Transfer, and Redial. On the right of the system, top side are the 4 way selection pad with select and delete, then below that are Menu, Messages, and Do Not Disturb, and finally Hold. In the middle are the 12 keypad keys, 4 soft keys, and volume + and - buttons. No where am I able to find a hard voicemail button. -Chris On 10:42 AM 7/19/2004, Wiley E. Siler wrote: Thank you! Can you tell me more about the dial plan feature? How do you setup the correct digitmap? W -Original Message- From: Russ Beaupre, P.E. [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 4:56 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Polycom IP 500 Voicemail Wiley E. Siler wrote: I have a solution that allows me to assign a soft key with no problems. However, it seems like a waste the the hard button labeled Voice Mail is not dialing right into voice mail. Is there a known way yo do this? I have tried everything in the manual but it doesn't seem to work. I have IP 500s and I want to be able to use all three display lines for just lines on the phone. I think that feature is inly available on the 1.2.0 sip firmware. It works on ours but when you press it, you still have to pick a line, then connect. The line button goes right to the voicemail. Also, do you know if it is possible to program the buttons along the bottom of the screen like normal soft buttons? Probably, but I haven't looked into it enough And finally... Is there a way to make the system dial without having to hit the Send key after dialing a number? look at the digitmap in sip.cfg -rb ___ Asterisk-Users mailing list [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] Rotary phones? (No, I'm serious)
I don't know how serious pulse dial is, but it is supported in asterisk. You will not likely find any device that converts pulse to tone though. Although it might be possible if it went through a channel that doesn't use pulse dialing like sip. Hp So a channel bank w/ FXS ports will pass the rotary data thru to Asterisk? Neat. ___ Asterisk-Users mailing list [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] Polycom IP 500 Voicemail
On Mon, 2004-07-19 at 12:42, Wiley E. Siler wrote: Thank you! Can you tell me more about the dial plan feature? How do you setup the correct digitmap? It is all in the Admin Guide you can download from the Polycom web site. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
From: Christopher L. Wade [EMAIL PROTECTED] Organization: Unistar-Sparco Computers, Inc. To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration Reply-To: [EMAIL PROTECTED] Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20, I don't think I'll need the U30, but I'm not entirely sure. The U20 will be fine. The U30 adds MF receivers for Feature Group D/E911. The T-1 is just set up as EM tie line. Good Luck Jason Kawakami Open Telephony Labs, LLC ___ Asterisk-Users mailing list [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] Cheap PoE switches/injectors?
You can pick up 3c17205's on ebay for usually around $500-$700 new in the box (non on there at the moment). They come up about 3-4 a month although it's summer at the moment so a bit quiet. P -Original Message- From: Scott Laird [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004, 11:42 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cheap PoE switches/injectors? On Jul 19, 2004, at 9:03 AM, [EMAIL PROTECTED] wrote: Look out for 3c17205 switches from 3com and read the QOS thread posting here at the moment. So $1600 for 24 ports. That's not *too* bad. HP seems to have a similar model (2626-PWR) for a similar price. 3com also seems to have a 24-port injector for $800. Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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: Cisco 7960 SIP V6 and distinctive ring.
Hello. Here is what my extension which uses distinctive ring on a Cisco 7960 running V6.2 firmware looks like. Note that the distinctive ring tones are changes in cadence, rather than changes in ringing sounds on the 7960. Also, if you adjust the ringer volume wile the distinctive ring is sounding, the phone will revert to the non-distinctive ring cadence. -Brian } Hi } } Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? } } I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. } } Thanks in advance. } } P } ___ Asterisk-Users mailing list [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] Cheap PoE switches/injectors?
On Mon, 19 Jul 2004, Kevin P. Fleming wrote: Scott Laird wrote: So $1600 for 24 ports. That's not *too* bad. HP seems to have a similar model (2626-PWR) for a similar price. 3com also seems to have a 24-port injector for $800. I still don't understand why I can buy single-port injectors for $20, but multi-port models are $30 per port and up. You'd think that having a single combined power supply and other bits would reduce the cost, not increase it. officially, a POE capable switch/etc is supposed to do a discovery routine to detect, when a device is plugged into it, whether that device requires POE. Right? And the single-port POE injectors are usually nothing more than two RJ45 packs with a dc power connector, right? That could be the difference in price there: the detection circuitry. Or am I way off? Greg ___ Asterisk-Users mailing list [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] collect calls
Hi, Does anybody knows where can I change timing for collect calls? tks Oz _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ Asterisk-Users mailing list [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] Flash Zap trunk from a Sipura
I think this has been an ongoing issue. When you figure out a solution let me know. The only solution I could come up with isnt feasible for everyone. I have another pbx that provides the dialtone for my asterisk box. I put two zap cards in my asterisk. On my other switch I set it so that if line 1 is busy it rolls to line 2, this way when my second call comes in I can switch over to it from my Sipura. It works for me, but isnt possible for everyone. On Mon, 19 Jul 2004 10:09:30 -0700, Trevor Peirce [EMAIL PROTECTED] wrote: Hello, In my quest to create several proof of concepts for what can be done with Asterisk, I've run into a bit of a problem. I have a pair of SPA-2000's acting as off premise extensions for an analog line. When a call waiting call comes in, the caller id information makes it though the ULAW codec and displays on the caller id box, however asterisk doesn't seem to want to pick up the hook-flash sent by the Sipura to answer that second call. I have configured the Sipura to send hook-flash messages to asterisk, and it does, but asterisk doesn't seem to know what to do with them. I've searched the wiki and google with no success. Thanks, Trevor ___ Asterisk-Users mailing list [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] Polycom IP 500 Voicemail
I read the administrator document repeatedly. I have not been able to find a wiki that applied to digitmap feature at all and I have searched repeatedly and read several of the wikis regarding Polycoms. The administrators guide doesn't have enough context explanation to make the use of the digitmap understandable. That is the basis of my request for a digitmap explanation. I am not asking someone to write mine for me. I am asking to see an example and an explanation that gives context so I can write my own and know I have done it properly. My PBX is Asterisk and the setup is about as generic as generic can be. Polycoms over SIP to the PBX. If you know where the wiki is for digitmaps please send it. If you feel inspired, a short explanation of the relevance and context of digitmaps would be greatly appreciated. I know everyone has to take their own time to answer these emails and I truly appreciate that. That is why I do my research until I hit a wall, then I will ask here. I appreciate whatever you can spare time for. Thanks! Wiley -Original Message- From: Brent Franks [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 10:26 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail Thank you! Can you tell me more about the dial plan feature? How do you setup the correct digitmap? Check the Administrator's Document. You can find it on the Wiki, under IP Phones.. Polycom. Did you try to look up the digitmap feature before sending this post? If not, you should be able to understand it when you read it, it's relatively straight forward. No one can setup a correct digitmap for you, as it will vary greatly on how you have setup your PBX. - Brent ___ Asterisk-Users mailing list [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] Mac OS X installer: missing files fix
Wallingford, Ted wrote: (B (BI've paraphrased the OS X installer developer's comments: (Bthere's a (Bbug in Installer that is preventing the archive from (Bworking right. (BBelow is the fix for the problem. (B (BApple may have a reputation for attention to detail and (Bperfectionism, but their PackageMaker utillity -- which is (Bwhat you use to create those install packages -- does most (Bdefinitely NOT share any of these virtues. It's one of the (Bworst examples of sloppiness I have seen. (B (BPackageMaker simply refuses to include files targeted at (B/usr (and below) into the bills of materials file (B(Archive.bom). The files are all there, nicely (Bshrinkwrapped into the archive itself (Archive.pax.gz), (Bbut no matter what you do, they won't show up in the BOM (Bfile. As a result, the installer will not install but (Bignore them. (B (BI am now going to change the target to /private/tmp and (Bthen run a postinstall script (luckily this feature (Bactually works) to move the files into /usr. I will also (Bprovide a patching utility for those who have been hit by (Bthis. (B (BFolks, I am very sorry about the inconvenience. I have (Btested the installer on various systems beforehand, but I (Bmust have missed to wipe everything at some point. My (Bsincerest apologies. (B (BFirst (obviously) run the installer. Since the (Bexecutables are in the (Barchive.pax.gz file in the installer package, first do a (B"show package (Bcontents" on the package file, then unstuff the enclosed (Barchive.pax.gz file to the desktop... Then open up a (Bshell, (BCD to the desktop, and run the following: (B (B cat Archive.pax | pax -r (B sudo cp -R usr/* /usr (B (BThanks for your follow up. (B (BYes, this will work, but I guess that it is a bit too (Bcrude for many Mac folks, they like to just click on (Bthings to make stuff work. That was the whole point of (Bcreating the install package in the first place. I have (Bfailed the Mac folks miserably in this regard. (B (BSo, please hang on there for a little while, I'll get back (Bto you with a new install package and a patching utility. (B (BMeanwhile, we've released a few clickable AppleScript (Bscript apps for basic control of Asterisk (B(start/stop/reload/show version). You can download them (B(as a zip archive) from (Bhttp:/www.astmasters.net/stuff/AsteriskApplescripts.zip (B (Bregards (Bbenjamin (B (B-- (BSunrise Telephone Systems Ltd (B9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Shibuya-ku, Tokyo, Japan (B (B__ (BDo You Yahoo!? (Bhttp://bb.yahoo.co.jp/ (B (B___ (BAsterisk-Users mailing list (B[EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP 500 Voicemail
And it is throughly convoluted in the admin guide. What is the T for? Pipe obviously separates entries. X = any digit one would assume? I am just luooking for a brief explanation. Thanks. Here is the excerpt from the manual. Attribute dialplan.digitmap Permitted Values string compatible with the digit map feature of MGCP described in 2.1.5 of RFC 3435. String is limited to 512 bytes and 20 segments; a comma is also allowed; when reached in the digit map, a comma will turn dial tone back on. [2-9]11|0T| 011xxx.T| [0-1][2- 9]x| [2-9]x| [2-9]xxxT Default Interpretation When this attribute is present, number-only dialing during the setup phase of new calls will be compared against the patterns therein and if a match is found, the call will be initiated automatically eliminating the need to press Send. Attribute Permitted Values Default Interpretation -Original Message- From: Eric Wieling [mailto:[EMAIL PROTECTED] Sent: Monday, July 19, 2004 11:50 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Polycom IP 500 Voicemail On Mon, 2004-07-19 at 12:42, Wiley E. Siler wrote: Thank you! Can you tell me more about the dial plan feature? How do you setup the correct digitmap? It is all in the Admin Guide you can download from the Polycom web site. -- Eric Wieling * BTEL Consulting * 504-899-1387 x2111 In a related story, the IRS has recently ruled that the cost of Windows upgrades can NOT be deducted as a gambling loss. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [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: Cisco 7960 SIP V6 and distinctive ring.
[Try this again...] Hello. Here is what my extension which uses distinctive ring on a Cisco 7960 running V6.2 firmware looks like. Note that the distinctive ring tones are changes in cadence, rather than changes in ringing sounds on the 7960. Also, if you adjust the ringer volume wile the distinctive ring is sounding, the phone will revert to the non-distinctive ring cadence. -Brian exten = 2135551212,1,setvar(ALERT_INFO=4) exten = 2135551212,2,Dial(SIP/100SIP/401SIP/403|20|tr) exten = 2135551212,3,Voicemail,u401 } Hi } } Can anyone with distinctive ring on their 7960's possibly post how they've got it to work? } } I understand that the ALERT_INFO variable is involved but using the examples for the variable value from the WiKi I'm just getting an error message from the Asterisk concole. } } Thanks in advance. } } P } ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
