RE: [Asterisk-Users] ISDN card required
So far - the 4-port ISDN HFC chipset cards from Junghanns.net works well and is half the price of a 4-port Eicon card. On Mon, 2005-11-14 at 10:07 +, David Waugh wrote: Hi Lee, I use a Diva Server card here with Asterisk using Chan_capi. The basic BRI card has one BRI port. They also have a model with 4 port BRI model. You can mix and match Diva Server card too, so as your needs expand you can add more cards to your server. Further information can be found on the Eicon website: http://www.eicon.com/worldwide/solutions/Diva_Server_and_Asterisk and http://www.eicon.com/worldwide/products/MediaGateways/all-in-one.htm Thanks David -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lee Archer Sent: 14 November 2005 09:32 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ISDN card required Can anyone point me in the direction of a quality, works with Asterisk, BRI card. I need minimum 2 port/4 channel. Regards Lee ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anybody tried it from India ?.
I can not see that its illegal to have Asterisk in India. The TDM400P card should work fine - but it may not be approved to be interconnected to the phone system. (This never stopped me doing similar things). I'm assuming that its possible to connect a 2-wire phone to the Indian phone system - ie - if you have ever bought a 2-wire phone from the USA and got it to work - then there should be no problem. In the UK - BT use a 3-wire system, the extra wire for ringing a bell... but actually provide 2-wire to the house. People seem to have little difficulty with the TDM400 there. I've had no problems all over Africa - you should be fine. Asterisk makes a great (cost wise) and highly functional PABX replacement. This in itself is reason to install Asterisk. The fact that it does VoIP as well is an additional bonus - just don't get caught using it? Up until the beginning of this year, VoIP was illegal in South Africa - never stopped most people. It is possible for telco's to monitor and even recognise and record 'voice' on the internet - but they usually look for common Codecs (u-law, a-law) and probably have better things to do. On Mon, 2005-11-14 at 15:50 +0800, Dinesh wrote: Its illegal to interconnect it to the local pstn (from abroad). Dinesh. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Rymes Sent: Monday, November 14, 2005 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anybody tried it from India ?. On Nov 14, 2005, at 12:37 AM, ram wrote: Hi its not legal in india connecting to PSTN to VOIP ram Asterisk doesn't necessarily mean VOIP. He could set it up using ZAP channels only and not have any VOIP in use at all. Tom Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: This email is confidential and may be privileged. If you are not the intended recipient, please delete it and notify us immediately. Please do not copy or use it for any purpose, or disclose its contents to any other person as it may be an offence under the Official Secrets Act. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] URL Dialing from SNOM phone
Couldn't find anything on the lists or in Wiki..
Customer wants to be able to dial complete SIP URL's... from his SNOM
phone.
ie - He dials on his phone [EMAIL PROTECTED] (which is more
difficult than a Number - but not undo-able)
How do I configure my extensions.conf to handle this sort of call?
I do have (which works!)
exten = 312,1,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED])
..in extensions.conf
Thoughts
1 - Treat it as a VOIP call, if it fails - Tough.
2 - Need a Extension rule like..
exten = [EMAIL PROTECTED],1,Dial(SIP/${EXTEN})
(which does not work)
3 - If it fails, playback Sorry, the URL you dialled can not be reached
Help anyone?
(I can see this type of call being made more frequently - ie to get my
support department - calling [EMAIL PROTECTED] via sip rather than
via e-mail..)
--
. . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
___
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Re: [Asterisk-Users] GSM cards / mobile phone cards for Asterisk?
On Fri, 2005-10-28 at 14:26 +0200, Tomasz Chmielewski wrote: So the idea is to put a SIM card inside the Asterisk box, equipped with a special card, a card which would be a mobile phone really. Does anyone have an idea if such cards exist, and if so, if they work with Asterisk? You can get Fixed Cell units... basically a Cell Phone which provides a Trunk line instead of screen and keypad. This looks then like an analogue trunk line. I believe that there is an Italian PCI card that has 4 cell units built into it. I believe that such units can also plug into an Ethernet and run SIP. or - Wait for the Sony Ericsson P990i cell phone which comes with Wifi - and stick on a SIP client.. and run wireless. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SNOM Subscribe/Notify
I'm using a SNOM 360 with Ver 4.3 software. Asterisk is Asterisk CVS-D2005.05.02.22.00.00-05/04/05 (BRI Stuff + Head) I've used the wiki info to set up some lines to monitor some internal extensions. When the extension is rung - the lamp comes on, when the call is answered, the lamp goes off.. I was expecting something a little more exciting - like the lamp to flash when the extension was ringing and for the lamp to go on when the extension was busy - either incoming or outgoing calls. Am I missing something here??? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] updating display of a hardphone based on agentslogging in
I'm also using SNOM320/360 phones. Ideally - set up one button to toggle the Agent Status (in/out == On/Off) ??? Kinda make sense if app_devstate (or similar) made it into mainstrean Asterisk - so line indication lamps could be used at will. The SNOM320 is so ideal for Call Centres (the Headset control it gives one) - I'm surprised that there is not a dedicated Agent Has Logged-in icon... :-) On Fri, 2005-08-26 at 10:20 +0200, Nils Ohlmeier wrote: On the Snom phones you can use a SIP MESSAGE to overwrite the idle screen text with a given text message. Maybe that is helpfull for your scenario. Regards Nils Ohlmeier Nils (or anyone else) - how does one do this from Asterisk? You've got the Snom 320's, so maybe the most straight forward thing to do would be to use the Hint application with them to light a status LED when an agent is logged in and have it go dark when the agent is logged out. We are settng up a fair sized call center on Asterisk, but we are having -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Queue/Agents
On Mon, 2005-08-01 at 18:31 -0400, Joseph wrote: Hall, Eric M. wrote: Looking for a good web app that will show agents that are login to queue. I tried the operator panel but I'm unable to get the LED to change color per the doco I have.. It works well for everything else but no luck on the agent part.. I can share mine. Shows a list of callers and agent status. OOh... please share... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] User authentication and privileges
I want to authenticate a user before he is able to use the phone. I also want to set his privilege as to where he is allowed to call to... Preferably, the password should be their VoiceMail password, (every extension (or is that user?) can have voicemail defined - even if its not in use?) ...one should be able to enter the password (variable length) as part of the dial sequence - eg the number to call is 0113140077 and the password is 1234 so dial something like *1234*0113140077 (no prompting!) and what should be written to the Accounts file should rather be the extension that that password is good for... (effectively - the User). This way, using voicemail.conf, users can manage their own passwords. I've seen some wiki stuff on AGI's that allow one to glean for user passwords.. If the system is smart (and the user not so), after dialing a trunk that needs a password and none were provided - then asterisk can prompt for it. It would also be cool if certain extensions did not need a password... (phone in MD's office?, Switchboard, Fax (maybe)) - this needs a flag against the extension - which could be a Privilege Flag. Privilege Flag: (suggestion) 0=internal calls (and emergency/911) 1=local calls 2=long distance 3=cellular 4=no barring at all (international) (Somehow need to Tag the class (privilege level) that a number falls into) Then what about an additional field in the voicemail.conf file that specifies what privilege a person has - ie from a phone with zero privilege, a user with priv 4 can use his password to make an international call... I say user rather than extension because a user should be able to call from any extension with their own password - the user has the restriction - not the extension. Anyone got anything like this? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GXP2000 and Headsets, Call Center phones.
I see the GXP2000 has a headset socket. Are their any compatible headsets for it. How does the functionality change? What else would people suggest for a Call-Centre? Would like Headset, Call Details - etc... The call centre answers the phone according to which number is called.. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] howto on ISDN HFC cards with AAH v1.1
On Sat, 2005-07-16 at 16:47 +0200, Zoltan Szecsei wrote: Hi, Can anyone please point me in a direction as to how to set up these 2 pci cards with AAH 1.1? Rather load [EMAIL PROTECTED] 1.3 - fixes other problems I have (am still) googling left, right center - but haven't found a definitive guide yet. The centos based setup lacks any of the tools I know (insmod, modprobe ) so it is time consuming just to even dig around the AAH box. There are no zaptel.conf files and on it goes. In 1.3 - I see... (/etc/asterisk) zapata_additional.conf zapata-auto.conf zapata-auto.conf.bak zapata.conf zapata.conf.template -and- /etc/zaptel.conf /etc/zaptel.conf.bak /etc/zaptel.conf.template There is still no auto-install for HFC cards though... The install-AVMB1ISDN (install support for AVB B1 ISDN card) does install some www.junghanns.net stuff but not HFC.. Soon maybe? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Difference between Asterisk and [EMAIL PROTECTED]
My 2c worth... For the beginner, AAH is great. The PC that you install on will be totally reformatted / fdisk-ed (assuming single drive - etc). With AAH 1.3 - the installation goes to sleep and sort of finishes when its Syncing with a Time Server. A reboot at this point seems to do no harm. As Asterisk is configured via AMP - you are limited in functionality as to what AMP can do for you - but one can edit the config files directly as well for custom configs. (I needed to program an incoming (Fax) zap line to go to one particular extension) As it starts - there are a number of dialplan features which are quite cool, eg Time, Weather, Wakeup-Call, You extension is.., VoiceMail, IVR, Do-Not-Disturb, FAX handling. Sure - these are all things Asterisk can do, but with the default asterisk download, you start with a pretty clean slate... My current AAH limitations include:- a) In IVR, no ability to program or hold for an operator timeout for the DTMF challenged. b) Support for junghanns cards (or HFC cards) c) Multi-Company support - Default is one primary IVR Just did an install with many extensions 16 lines (4 x TDM400P) - 2 to Fixed line Cells, 14 to Telco (no services except DTMF dialing - its in East Africa). -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT (kinda): Justification for adding Asterisk to the business plan
On Fri, 2005-07-15 at 04:17 -0700, /dev/null wrote: I'm trying to build a justification case to get the firm I work for to start working with Asterisk more. How could I build this case? The argument I'm raising is that people need phones. PBX systems are too expensive for fewer options and less expansion capabilities. Leveraging Asterisk in the business plan would allow for more consulting revenue and resale opportunities. As their business grows or reconfigures, the call back to reconfigure mail boxes and install additional IVR into the system would allow for better work flow and more interactive customer service. Problem is, they simply cannot see how Asterisk can fit into a normal IT consulting business plan as it's a Telephone thing and not a IT thing. Features I'd not ignore IVR units for PABX's - usually very limited and expensive, with Asterisk's ability to have multi-level IVR - This can be used as a front end to a customer ticketing system. Voice Recording - usually prohibitively expensive (or junk, ie a Tape Recorder with phone Mic) - ideal to recall what was actually said between Support staff and Customer Caller-ID - most analogue phones don't do this, with Asterisk - its almost a given - which helps identify the customer and give Support an edge. Can be extended to sent non-paying customers straight to Accounts. The fact that Asterisk is soft and you're trying to sell to an IT Company.. (Even) Installing [EMAIL PROTECTED] gives the ability to do most of the above - the Customer should have no issues with IVR Setup, Adding extensions - etc. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GUI
On Fri, 2005-07-15 at 17:36 +0300, [EMAIL PROTECTED] wrote: I was wondering which would be the best GUI to use for Asterisk management? astGUIclient or AMP? I'd use AMP - mainly because [EMAIL PROTECTED] uses it - so the user base and knowledge base should be bigger... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Do I need a ring capacitor to use TDM400P cards in UK
On Wed, 2005-06-08 at 23:39 +0100, David John Walsh wrote: Angus Jumping in with both feet a BT socket with a capacitor in is commonly refered to as a Master socket, and are very cheap even without wholesale. It gets its name from being the socket that BT installed into the house for the line, all other sockets in the house will be slave or secondary (ie no capacitor) (and its against the law to play with the one BT installed - but thats off topic!) ..and it complicated my understanding of how to get ADSL working at the same time so ADSL filter was installed before the Master... UK Phones at homes historically had a separate bell - mounted in the hallway. Phones where then placed where convenient. This allowed one loud bell (sucking current) and multiple (quieter, less current thirsty) phones... To do this, the 2-wire line from the Telco was altered into a three wire line inside the residence, the job of the 'Master Jack'. This is done with a capacitor from one of the legs to provide the third wire. Look inside the 'Master' to confirm... (there might also be a resistor from the other leg to the new third leg too). (I can remember playing with a crystal radio set, that needed an earth, and the instructions saying to use the metal (silver coloured) finger stop on the rotary dial as an earth - so there may be an earth as a fourth wire...) Because of this - many phones sold in the UK will only ring via this third wire... I vaguely remember bringing a cordless phone from the UK to South Africa (where the US 2-wire equipment work fine) and adding a capacitor inside the phone to make it Ring... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 4 port BRI options ?
On Fri, 2005-06-03 at 06:28 -0700, Nardis Dome wrote: --- Brett, Gary [EMAIL PROTECTED] wrote: Is the Eicon that much better ? sorry, i have only experience with Eicon... maybe someone else is able to give a feedback... I'm using Junghanns 4 port card. There is also an 8 port card. Installation is very simple, download a startup image from Junghanns.net and it does the rest... It works - I've no complaints. However - you are somewhat reliant on Junghanns.net for all future changes - etc. I'm running... Asterisk CVS-D2005.05.02.22.00.00-05/04/05 I'd be MUCH more comfortable if somehow Junghanns changes were rolled into the main stream code... I'd also love to see Digium with a multi-port BRI-ISDN adaptor for both US and non-US use. Its also possible to configure the card jumpers (add some power?) and then plug an ISDN phone into the card. ps - FAX reception works - as part of asterisk. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream GSX-2000 - dead :-(
I have a Grandstream GSX-2000 with .. Software Version:Program-- 1.0.0.3Bootloader-- 1.0.0.3 I tried to do an HTTP update from the Grand Stream web site... After half an hour, I recycled power and now its dead... LED's come on and stay on, screen and buttons are dead. Connectivity to Grandstream.com was always good - whenever I checked (I downloaded the User Manual in a couple of minutes), the site states five minutes to load, so waiting more than 30 mins should have been OK, and they do have this Please Powercycle in red print too... Is there a magic re-incarnation routine ? (Power on whilst holding down some buttons?, Sprinkling chickens blood?) I chose an HTTP upgrade over TFTP - as I read that there were potential issues with TFTP at this firmware level. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX to FWD?
On Thu, 2005-05-12 at 12:40 -0600, Tim Pushor wrote: I had trouble calling people who were using FWD/SIP from my FWD/IAX account. I switched back to using SIP and could call SIP users, but not IAX users. I've since de-registered myself for the IAX *beta* and can now talk to everyone again. I noticed something similar. My Asterisk box just uses FWD:SIP. I have two hardware capable IAX phones, couldn't get SIP to work on them (NAT problems) so tried IAX which worked fine from the phones to my box, but they could not call each other.. The IAX phones are in different countries/continents on different ADSL services (Parents, Brother - etc) Assumption - two IAX devices both registered at FWD can not talk to each other. ps - how does one set up a Proxy - so machines on foreign NATs can talk? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_yyerror - 'space' in Caller-ID - string comparison
I've some code to manipulate incoming Caller-ID - so its suitable for
replying to...
[sipdef]
exten = s,1,NoOp(FWD SIP: ${CALLERIDNAME} ${CALLERIDNUM})
; Alter incoming calles from pulver - add a '87'
exten = s,2,Gotoif($[${CALLERIDNAME} = ${CALLERIDNUM}]?3:4)
exten = s,3,SetCIDName(87${CALLERIDNUM})
exten = s,4,SetCIDNum(87${CALLERIDNUM})
exten = s,5,Goto(default,s,1)
When Executing the above - and I presume incoming Caller Info looks like
the name is Mark Elkins and the Number is 638936...
The purpose is to prefix the number (only the number) with 87.
Sometimes, incoming CallerID data looks like -- 638936 638936
therefore the checking of both Name and number.
-- Executing NoOp(SIP/292951-b11f, FWD SIP: Mark Elkins
638936) in new stack
May 12 14:36:59 WARNING[28824]: ast_expr.y:486 ast_yyerror:
ast_yyerror(): syntax error: parse error; Input:
Mark Elkins = 638936
^
-- Executing GotoIf(SIP/292951-b11f, Mark?3:4) in new stack
-- Goto (sipdef,s,4)
-- Executing SetCIDNum(SIP/292951-b11f, 87638936) in new stack
-- Executing Goto(SIP/292951-b11f, default|s|1) in new stack
-- Goto (default,s,1)
What solutions are there to getting rid of the yyerror??
--
. . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Re: [Asterisk-Users] good bri card not junghanns
On Fri, 2005-05-06 at 14:29 +0200, Eugenio De Vena wrote: Hi there, will someone suggest me a good and * combatible isdn card ( 1 , 2 , 4 , 8 channels ). I am currently working with but can not stand their complete lack of support. In all fairness to Junghanns, my current release Asterisk CVS-D2005.05.02.22.00.00-05/04/05-18:22:14 - is rather cool Someone was busy over the May 1st long weekend. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callto: URL (URI) tag for dialing
I see that there seems to be a 'callto' URL/URI for dialling a phone number... ie - on my web site's Contact Page - I have added the code... a href=callto:+27128070590+27 12 807-0590/a There should be some generic way for Mozilla (firefox - etc) to somehow turn a click on such a link into persuading Asterisk to dial the number for me and connect it to my SIP hard-phone. 1 - mini application under mozilla to collect the number/sip address, add in a static local extension (personal settings?) and pass info to a listener (auto-dialer) on the Asterisk Machine 2 - Auto Dialer dials my extension, then on answer, dials the URL or phone number. The URL could either be a simple phone number or a full SIP address?? Anyone done this? ..and care to share? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MozPhone
On Wed, 2005-03-02 at 07:23 -1000, Jean-Denis Girard wrote: Is anyone using mozPhone? If so any feedback you can provide? Yes. For what I'm doing with it work. Could be improved. Thanks for your feedback. MozPhone could obviously be improved in many ways, what would be your suggestions? Hi, Just tried your app today. I understand everything is intuitive to use ... but... have you released any sort of manual? Is the source code available anywhere? I love the extra's like the Manager Console - but the information is gives is sometimes a bit out.. (What Extension is '[EMAIL PROTECTED]:1' ??) Be nice if the Manager Console could remember settings (size/location - etc) and hang around even if the 'dial' interface closed. Still have not worked how to get rid of GnuMeeting when I click on a 'callto' link. I have not seemed to get the app to come up on a 'tel:' link either. Rather than using the PC's phone capability - I'd like to have the system use my HardPhone - can this be another option? - something to add to the config menus? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI: received SETUP message for call that is not a new call, wicked!
Hi list, I'm getting the message... Apr 4 15:13:09 WARNING[1069]: chan_zap.c:7512 zt_pri_error: PRI: received SETUP message for call that is not a new call, wicked!!! This is running Asterisk 1.0.6-BRIstuffed-0.2.0-RC7k. These messages happen when someone calls from the Telco on a BRI line... but rather than asterisk simply immediately answering, they just hear ringing So really the new call IS a new call - but Asterisk things differently. Anyone met and/or solved this problem? This seems to be happening to 1 in 4 of all my calls??? - other calls are fine. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work with asterisk!
On Thu, 2005-03-24 at 10:50 +0100, Marc SCHAEFER wrote: On Wed, Mar 23, 2005 at 06:55:28PM +0200, Mark Elkins wrote: Last time I tried - there were a few problems... I had a few random crashes, higher delays and echo with the EICON. I replaced it now with an HFC. The EICON on isdn4linux was however a bit better than the AVM C4 with CAPI. I am still curious. Which Driver do you use for the HFC card? It could be: bristuff-0.2.0-RC7k stuff from http://www.junghanns.net/ - but this locks you into using a particular - non-HEAD version of Asterisk.. (and missing all the new goodies) ..or something else (please tell..) I wish there were single, four and eight port ISDN BRI cards that Digium sold and supported - so I could run whichever version of Asterisk I wanted... This is because DTMF detection and sending is disabled in chan_modem. Ouch - wish I'd known this a few months ago... http://www.marko.net/asterisk/archives/0301/0849.html. http://lists.digium.com/pipermail/asterisk-users/2003-June/014104.html -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon DIVA PCI ISDN cards (not server) work with asterisk!
On Wed, 2005-03-23 at 17:18 +0100, Tomasz Chmielewski wrote: I just wanted to let you know that it's possible to use Eicon DIVA PCI 2.01 ISDN cards (not server divas) with asterisk. Last time I tried - there were a few problems... 1 - Outbound DTMF - never made it... ie You can not interact with someone else's IVR (DTMF controlled systems) 2 - Inbound DTMF - Certain voices would be interpreted as DTMF - which is fine until they sounded like a '#' - and got transfered (some strange reason - my wife's voice - especially when she got angry) I believe that there was some sort of patch for (2) but never heard of a fix for (1) Has this changed at all??? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF out to Cell Phone
On Tue, 2005-03-08 at 14:16 -0500, John Fullington wrote: I set up a monitoring system that calls my techs when a problem occurs on one of our networks, everything works fine unless asterisk calls a cell phone in which case the tech can not respond using dtmf. It works fine if the tech call in but not if asterisk call a tech's cell phone. Anyone one have any suggestions? The application sounds interesting. Any chance you can email more about what you are actually doing? (code?) It sounds like your problem has nothing to do with mismatching Codec's or how the DTMF is being sent... etc... I have an Asterisk installation with BRI and with a premicell attached to an analogue interface (Premicell=fixed cell phone with analogue 2-wire interface that gives dial tone - like a trunk line) Perhaps I can then confirm your problem - or help with a solution? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phoneproblem
On Fri, 2005-02-25 at 16:29 -0700, Mr. James W. Laferriere wrote: Hello Mark , C. All , Is this device available for sale in the US ? All the digging I've only found outside US mentions of sales . Any help appreciated . JimL No idea. The Unit I have is a locally manufactured device called Digi-Cell - frmaritz (at) global.co.za is the email address on the box it came in Its probably 900Mhz GSM only - in the US - You'll need a 1900Mhz unit??? On Fri, 25 Feb 2005, Mark Elkins wrote: On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phoneproblem
On Mon, 2005-02-28 at 17:50 +0200, Herman Cremer wrote: http://www.psitek.co.za/gsm.html These guys are also in RSA, and Australia. This unit does exactly the same as the DigiCell, which mark is talking about, but is a much better product (and more expensive) maybe they export ? Except that when I've been to the USA - I've needed a 1900Mhz phone - this is only 900 and 1800... *GSM INTERFACE * GSM output 900MHz: Class 4/5, 2W EGSM * GSM output 1800MHz: Class 1, 1W DCS * SIM interface: 3V mini SIM but look at the website (Hey, it looks like my box!) as the features are what you are looking for... I believe Motorola was one of the earlier producers of this type of device - but would think that most of the manufacturers would have a similar type of unit. Push the Telephone access in disaster areas, where wire-network infrastructure is damaged point... :-) -Herman On Mon, 2005-02-28 at 17:21, Mark Elkins wrote: On Fri, 2005-02-25 at 16:29 -0700, Mr. James W. Laferriere wrote: Hello Mark , C. All , Is this device available for sale in the US ? All the digging I've only found outside US mentions of sales . Any help appreciated . JimL No idea. The Unit I have is a locally manufactured device called Digi-Cell - frmaritz (at) global.co.za is the email address on the box it came in Its probably 900Mhz GSM only - in the US - You'll need a 1900Mhz unit??? On Fri, 25 Feb 2005, Mark Elkins wrote: On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. Content preview: http://www.psitek.co.za/gsm.html These guys are also in RSA, and Australia. This unit does exactly the same as the DigiCell, which mark is talking about, but is a much better product (and more expensive) [...] Content analysis details: (0.9 points, 5.0 required) pts rule name description -- -- 0.1 FORGED_RCVD_HELO Received: contains a forged HELO 0.8 CELL_PHONE_IMPROVE BODY: Talks about cell-phone signal improvement -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phone problem
On Wed, 2005-02-23 at 14:22 +0100, Roberto Piola wrote: We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10) and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are configured in TE mode and connected to the PSTN; the other 8 are in NT mode and connected to isdn phones. the other outbound calls to PSTN are fine, however, when we call cellular phones, often audio is one-way (i.e.: the cell phone user can not hear, while the speaker at the internal side hears perfectly. CPU usage is quite low, and asterisk -rvvv does not show anything particular In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel (Junghanns 4BRI card) to cell phoneproblem
On Fri, 2005-02-25 at 13:46 +, C. Tomlinson wrote: Did you have to make any changes to use the premicell, or was it as simple as an outgoing landline call? I am looking into doing this as you can get deals where calls between chosen numbers are free :-) Absolutely no changes at all I did stick a Phone onto the 2-wire input of the 'PremiCell' to check that all worked - before going via Asterisk - but thats all. [part of the previous message] In South Africa, I have a 4-port ISDN BRI (Euro-ISDN) with bristuff. Calls to Cell phones are no different to any other call... I also added a Digium 4-port analogue card - and have a 'PremiCell' connected to a Trunk line. The PremiCell is a fixed cell device that gives dial-tone in the same way that a Telcom Trunk line would work - except there is no copper to he exchange - just a stubby cellphone antenna. In South Africa it is MUCH MUCH cheaper to make a Cell to Cell call than from Telcom to Cell I'm surprised that more people do not put down a 'PremiCell' type device and route all Cell calls out through it... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Initializing two ISDN cards in isdn4linux
On Sat, 2005-02-12 at 12:20 +, JunkMail wrote: For the single card I was using with isdntool for initialization, wich works fine but has no support for two cards. Can anyone tell me exactly how to initialize the ISDN system manually ??? It all starts with modprobe -v hisax type=21,21 (loading hisax and telling it that we'll use two teles pci cards) and then ? what else ??? Not sure if this will help you - I ponce played with a mixture of single port cards... Can't remember where I got the 'id=' bits from.. # For one eicon PCI #modprobe hisax type=11 # For two eicon cards modprobe hisax type=11,11 id=201%202 # For Asuscom ISA # isapnp /etc/isapnp.conf #modprobe hisax type=12 irq=3 io=0x0100 -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP extn number planning
On Tue, 2005-02-08 at 06:27 -0600, Rich Adamson wrote: Looking for some advanced thoughts relative to exten number assignments. We're in the planning stage for rolling out asterisk at multiple small US telco/isp operations. Their typical voip customer has had their pstn line for a long time and wants to keep the pstn line and number, but add voip to their existing home/soho arrangement. The approach that I have taken is... 1 - at each place that I have asterisk, register the users full number with e164.org (or equivalent) 2 - Make sure I do e.164 lookups as part of the normal process of placing a call... 3 - If a call comes in via VoIP - alter its CLID so it looks the same as an incoming telco call - which makes identifying and returning the call simple. Effectively - I use the dialling plan from Telco. Each site retains its 'historical' number - which is probably the same as everyone has in their Rolodex/Diary/PDA (etc) - so there is no customer learning - or dialing funny access codes - etc If the call does not get through - my system simply uses the Telco line - as in the old way. If your client calls anyone else who implements the same rules - they'll get through on VoIP too... and if I take a phone book, look up your customer and call the number given - I'll use VoIP too... The only time that I do not do any number lookups is to 911 or operator specific numbers... which in South Africa tends to be '10XXX' format. This works fine for any Asterisk installation that has both traditional (= fixed connection to telco) and VoIP circuits. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P specs clarification
On Mon, 2005-01-31 at 15:30 +0500, [EMAIL PROTECTED] wrote: Hello, I need some clarification on TDM400P. The TDM400P card by itself has no use. You purchase a mix of FXS and FXO daughter cards (they are coloured Red and Green) which pug into four available positions on the card. That decides the functionality of the TDM400 card. In terms of FXO and FXS what does it mean. I can see that it has four RJ 11 sockets. How will you decide which of the four interface to use for what. I mean FXO or FXS. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.0.2-BRIstuffed-0.2.0-RC2b and '*8' calls dropping
I'm using Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b - when anyone picks up a call with '*8' - the call will drop after about 20 or so seconds. Is this a general problem with Asterisk 1.0.2? As this is the latest release that it appears Klaus-Peter Junghanns has for public consumption - is there anything I can patch for just this problem - or has Klaus-Peter Junghanns (or anyone else) been quietly busy with a BRIstuffed patch that works against Asterisk Head? I also notice that I can't seem to re-compile the H323 stuff any more... with this release... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream setup woe and solution
Just added a new Grandstream BT102 to my network. Its running new firmware (Ver 1.0.5.22 of 2005-01-21). I could NOT get the damn thing to (SIP) register Gripe 1: The New Firmware does NOT show the current version of all the firmware. You have to ask the phone manually with its menu button. Gripe 2: It does not show '' in the the two password fields... This is what caught me - I had two browser (tabbed) sessions and was switching between them - looking for differences... obvious the password fields now being blank look the same.. I never typed in the Authenticate Password: Doing so fixed the problem. If anyone from Grandstream lurks - can they change this behaviour? - at least fake some '***' in the password fields... Asterisk also had me chasing my tail - it never mentioned anything such as 'SIP Registration password is incorrect' - I got one.. chan_sip.c:7231 handle_request: Failed to authenticate user Phone Five sip:[EMAIL PROTECTED];user=phone;tag=fjhgkjhgkhjlk (OK - failed authentication - but something about the password would have been better) and got lots of... chan_sip.c:7588 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '192.168.0.126' ... which had me greping around for the word phone (should this have not been phone5 ??) -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream Bugetone 101 mwi
On Sat, 2005-01-15 at 09:09 -0500, Doug Lytle wrote: Mike Dent wrote: Whilst on the subject of BT's, do the callers and called buttons function? they dont seem to do anything on mine? Yes, but the hand set needs to be off hook. To add to Doug's reply... ---for people you have called--- 1 - Pick up phone (or push 'speakerphone') 2 - Push 'called' - keep pushing it again and again - the displayed number should change and the location where the time is usually displayed will also change (increment)... 3 - When you get to the number you wish to call again - push 'send' For people who have called you - exactly the same - except push the 'callers' button. The trick here is to make sure that the caller-id info that the phone has saved (the people who have called you) somehow can be sanely understood by your dial-plan logic.. I believe this works for the last 20 'called' and the last 20 'callers'. Only flaw in the logic is that it would be nice to push the callers/called button - select the appropriate number and then when pushing either 'send' or 'speakerphone' - activate the speakerphone and dial the number... -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream Bugetone 101 mwi
Curiosity got hold of me. I opened up my BT-10 (and it still works
afterwards..)
Under the keyboard (buttons) are four red LED's that appear to run in
parallel (they all flash at the same time when you put the power on).
These are used to light up the keyboard.
The Display LED (blue in my case) is flashed to indicate that a Message
is waiting
There appears to be no other LED's (or light sources) so no button will
ever (or can ever) flash...
In order to get the message button to work - programme it with the
extension number for your voice-mail. On your BT-100's phone web page -
it looks something like..
Voice Mail UserID:[300] (User ID/extension for 3rd party voice
mail system)
So if I push the 'Message' button - I effectively dial '300' (ie the
same as picking up the handset and dialing '300'). In my
extensions.conf file - the appropriate line is...
; 300 = Access Voicemail
; My 'Grandstreams' have a Message button - that I have programmed to
dial '300'
; They then pass over their CLID - so get to the correct mailbox
exten = 300,1,VoicemailMain(s${CALLERIDNUM})
exten = 300,2,Hangup
This will contact the Voicemail menu system - passing it the ID of the
phone that is calling it - the 's' is to skip the password
authentication..
Every BT-100 phone is set up in the same way - with the same '300' in
the Message Button field.
I also have the following set... to **YES**
SUBSCRIBE for MWI:
Yes, send periodical SUBSCRIBE for Message Waiting Indication
So, with reasonably new firmware - the only button that does not seem to
have a function is 'Conference'. The 'Transfer' button is used for
attended (non-blind) transfers (see postings elsewhere).
On Fri, 2005-01-14 at 23:47 -0700, Paul Fielding wrote:
Hahawell the MWI is the blinking blue LCD. The message button
is reserved for future use Hang in there. There will soon to be some
upgrades and rumor has it that the conferencing feature will soon be
introduced so that conference button on the phone will soon be
working.
The message button isn't reserved, it works fine, you simply need to
correctly configure it. It's job is to dial the voicemail box when
pressed. This works as designed. It just doesn't blink.
On Fri, 14 Jan 2005 10:25:46 -0500, Stephen R. Besch wrote
Ronald Wiplinger wrote:
I tried to use message waiting indicator, by Subscribe for MWI in the
web menu of the phone.
However, it does not light up / flash, even if a voice mail is waiting.
Where is the switch to turn it to?
I don't mean to be rude to everyone who responded to this question,
but I think that everyone is answering the wrong question. The
point is that the message waiting indicator doesn't light up, at all,
ever. All that happens when messages are waiting is that the
display blinks and the phone gives a stutter dialtone. That's it.
There is no light under the button - there should be, but there
isn't. The blinking phone designers should have put those stupid
blinking red leds - that only flash on boot up - under the message
button and flashed the display during boot up. But they didn't and
we're stuck with it. Such is life.
--
. . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Re: [Asterisk-Users] Displaying incoming e.164 callers number - how?
On Tue, 2005-01-04 at 15:34 +0100, Erik Versaevel wrote:
Mark Elkins wrote:
On Tue, 2005-01-04 at 15:45 +0200, Mark Elkins wrote:
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
Playing with myself again - that is - I called myself - and
got the caller ID of '27128070590'... not quite what I wanted...
In my extensions - I have...
[fromaix]
exten = 27128070590,1,Goto(default,s,1)
And again - changed the above to...
[fromaix]
exten = 27128070590,1,GotoIf($[${CALLERIDNUM:0:2} = 27]?2:4)
...
Default section looks like...
[default] ; what people will get when they call me.
exten = s,1,NoOp(CALLER=${CALLERIDNUM})
exten = s,2,Answer()
how about SetCallerId(12345) ;)
ie
exten= 27128070590, 2, setcallerid(0${CALLERIDNUM});
This works fine... Thanks.
Incoming AIX looks like...
[fromaix]
exten = 27128070590,1,GotoIf($[${CALLERIDNUM:0:2} = 27]?2:4)
exten = 27128070590,2,setcallerid(0${CALLERIDNUM:2})
exten = 27128070590,3,Goto(default,s,1)
exten = 27128070590,4,setcallerid(09${CALLERIDNUM})
exten = 27128070590,5,Goto(default,s,1)
... and does the right thing...
Of course - this depends on people making e.164+VoIP calls to me
actually setting their Caller ID according to the format '27128070590' -
ie - No plus signs (as for cell/mobile phones), no '00' (or other access
code for international dialling - just the country dialing code followed
by their whole dialing code...
--
. . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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[Asterisk-Users] Displaying incoming e.164 callers number - how?
I've got asterisk able to make and receive calls via the Internet via E164 lookups. If I get such a call - I'd like to display the original phone number on my phone. In the log is the following - which displayed '601' on my phone. The caller was +886288097680 - am I getting the wrong ClID because of my end or the caller end? ,601,3,default,601,IAX2/[EMAIL PROTECTED]:4569/1,SIP/phone3-99fb,Dial,SIP/phone3|30|tr,2005-01-03 16:53:33,2005-01-03 16:53:33,2005-01-03 17:02:00,507,507,ANSWERED,DOCUMENTATION Anyone care to call me? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Displaying incoming e.164 callers number - how?
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
phone number on my phone. In the log is the following - which displayed
'601' on my phone. The caller was +886288097680 - am I getting the wrong
ClID because of my end or the caller end?
,601,3,default,601,IAX2/[EMAIL
PROTECTED]:4569/1,SIP/phone3-99fb,Dial,SIP/phone3|30|tr,2005-01-03
16:53:33,2005-01-03 16:53:33,2005-01-03
17:02:00,507,507,ANSWERED,DOCUMENTATION
Anyone care to call me?
Replying to myself - I see that the gent at +886288097680 should be
doing something like
[macro-enum-call]
exten = s,1,SetCallerID(27128070590)
exten = s,2,EnumLookup(${ARG2})
... ie setting his own caller-ID before calling...
--
Playing with myself again - that is - I called myself - and
got the caller ID of '27128070590'... not quite what I wanted...
In my extensions - I have...
[fromaix]
exten = 27128070590,1,Goto(default,s,1)
..and..
[default] ; what people will get when they call me.
exten = s,1,NoOp(CALLER=${CALLERIDNUM})
exten = s,2,Answer()
...etc...
This gives me (in the console)
-- Executing Goto(IAX2/[EMAIL PROTECTED]:4569/2, default|s|1)
in new stack
-- Goto (default,s,1)
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569/2,
CALLER=27128070590) in new stack
-- Executing Answer(IAX2/[EMAIL PROTECTED]:4569/2, ) in new
stack
- so how do I rewrite the caller id - such that if it starts with '27' -
I change the '27' to a '0' - otherwise prepend it with (South Africa's
international access code) '09' ???
My main phones are all Grandstream's - and I'd like to be able to
uniformally return a call regardless of how it arrived
OK - so people calling via VoIP can fake (or simply never setup) their
caller ID - but I'm looking for utopia.
--
. . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Re: [Asterisk-Users] Displaying incoming e.164 callers number - how?
On Tue, 2005-01-04 at 15:45 +0200, Mark Elkins wrote:
On Tue, 2005-01-04 at 15:20 +0200, Mark Elkins wrote:
I've got asterisk able to make and receive calls via the Internet via
E164 lookups. If I get such a call - I'd like to display the original
Playing with myself again - that is - I called myself - and
got the caller ID of '27128070590'... not quite what I wanted...
In my extensions - I have...
[fromaix]
exten = 27128070590,1,Goto(default,s,1)
And again - changed the above to...
[fromaix]
exten = 27128070590,1,GotoIf($[${CALLERIDNUM:0:2} = 27]?2:4)
exten = 27128070590,2,SetGlobalVar(CALLERIDNUM=0${CALLERIDNUM:2})
exten = 27128070590,3,Goto(default,s,1)
exten = 27128070590,4,SetGlobalVar(CALLERIDNUM=09${CALLERIDNUM})
exten = 27128070590,5,Goto(default,s,1)
Default section looks like...
[default] ; what people will get when they call me.
exten = s,1,NoOp(CALLER=${CALLERIDNUM})
exten = s,2,Answer()
Logic flow is meant to be..
1 - if CallerIDNum starts with '27' - goto line 2 - else goto line 4
2 - Remove the first two digits off the CallerIDNum, replace with '0'
3 - Goto my default section - normal processing
4 - Prepend the CallerIDNum with '09'
5 - Goto my default section - normal processing
Problems -
The callerIDNum variable does not change :-( I thought that is what
'SetGlobalVar' was ment to do??? Seems to be ReadOnly or in a local
context... - how do I 'export' the change?
The Console shows...
-- Executing GotoIf(IAX2/[EMAIL PROTECTED]:4569/4, 1?2:4) in
new stack
-- Goto (fromaix,27128070590,2)
-- Executing SetGlobalVar(IAX2/[EMAIL PROTECTED]:4569/4,
CALLERIDNUM=0128070590) in new stack
-- Setting global variable 'CALLERIDNUM' to '0128070590'
-- Executing Goto(IAX2/[EMAIL PROTECTED]:4569/4, default|s|1)
in new stack
-- Goto (default,s,1)
-- Executing NoOp(IAX2/[EMAIL PROTECTED]:4569/4,
CALLER=27128070590) in new stack
--
. . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
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Re: [Asterisk-Users] SIP Phones-Receptionist Setup
On Thu, 2004-11-25 at 13:24 -0600, Carmi Weinzweig wrote: Again, note that I am not asking to display trunk status, just extension status, and to allow a user to place a call on hold on one phone and pick it up on another (that has that shared extension). From another posting today (SNOM telephones and LEDs) that should be possible (the status part). I'm waiting for the new Budge-Tone phones - that have LED's on Keys - in order to do this myself. I'm expecting to be able to show a couple of other status's - eg: a) Night Mode (Toggle with an adjacent Night Mode Key) b) Do Not Disturb (Again - Toggle with the adjacent button) c) Unconditional Forward-To (Toggle with adjacent button) I see all these as being pretty generic features - somehow interrelated to DataBase variables... Surely this is all possible. PS - My Grandstream phones (BT100) with 1.0.5.18, and Send-Flash-Event-as-DTMF=No, now are doing Attended transfer just fine! -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [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] Re: SIP Phones-Receptionist Setup
On Fri, 2004-11-26 at 09:05 +0100, hhandresen wrote: OT: http://www.grandstream.com/BETATEST/ (as someone else on this list stated) I've not seen any problems with it yet Sequence is, you have a call, push Flash, dial new extension - speak, push transfer - and you're out of the loop. But where did you get the 1.0.5.18 firmware ? PS - My Grandstream phones (BT100) with 1.0.5.18, and Send-Flash-Event-as-DTMF=No, now are doing Attended transfer just fine! -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [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] Grandstream GXP-2000
http://www.grandstream.com/VON_Fall_2004_Product_Announce.pdf talks about the new GXP-2000 - the replacement for the planned BT-102D (which I was waiting for) Anyone seen one yet? Anyone care to say anything about it - price, performance - etc... ...or should I look elsewhere... Been wanting.. 2x10/100, Power-over-Ethernet, AlphaNumerical Display, some line indicators extension buttons so the physical characteristics are there! If the software is similar to the BT-101 - then it should work OK? -- . . ___. .__ 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] H323 without a gatekeeper
Setting up a Gatekeeper can be a pain. After looking at Speed Dial /
New Context from Wed, 3 Nov 2004 18:24:31, I added the following bits
into 'extensions.conf'.
Maybe useful to others..
In my incoming default profile - I have...
; Calls from the H323 Extentions
exten = s/205,1,Macro(h323extn,Mark)
exten = s/206,1,Macro(h323extn,Alistair)
-and-
[macro-h323extn]
exten = s,1,Read(ToDial,posix/no2call) ; please enter the No. to call
followed by #
exten = s,2,GotoIf($[${LEN(${ToDial})} 2]?sip,${ToDial},1:3) ;Valid
number length?
exten = s,3,Hangup
On my Planet H323 phone, I have programmed a key so it simply calls
Asterisk - and looks like an external incoming call. Caller-ID from the
phone does a simple (unsecure!) validation (I'm using 205 and 206 on two
phones). I'm using Macro's because I was doing some odd things before.
In the Macro - the LENgth check is to cleanly get rid of unwanted call
attempts. The user-requested number is then simply fed back into the
context I use for my Sip phones (sip).
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
/ |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496
IN my incomming default
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Re: [Asterisk-Users] Grandstream and CallerID
On Wed, 2004-10-27 at 08:15 -0500, Eric Wieling wrote: George Gardiner wrote: I would be grateful for any pointers in the right direction. In short, I get CallerID to display on Xten and a SipTone II; but have failed miserably to get my BudgeTone 101 to display anything other than the phone's own number. The BT101 can only display callerid number. It's a number only display. Not quite - when someone calls from out of the country (no caller ID) - then the BT100 tries to display'Trl' or something like that... -- . . ___. .__ 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] Bri solution for Asterisk
On Wed, 2004-07-21 at 16:55, Massimo De Nadal wrote: I'm using a Cologne chip card in my Asterisk box with zapHFC drivers (bristuff-0.0.2). The system works well, but this way I'm not able to run newer version of Asterisk. Do you think it's better to use i4l support and newer version of Asterisk or keep the bristuff with older asterisk ?? going to i4l means... incoming sound sometimes gets interpreted as DTMF - and when your caller humms a '#' - transfer kicks in... Outgoing DTMF simply does not work. (Don't do i4l!) There is an Update patch for bristuff... look carefully in the download directory. Have anyone tried chan_mISDN on a 2.6 box ? How does it run ??? Dunno - try it and let us all know. -- . . ___. .__ 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] 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: AW: [Asterisk-Users] Junghans Quad-BRI card and asterisk cvs-head
On Tue, 2004-07-06 at 11:29, Martin Bene wrote: The bristuff distribution comes with a install.sh script (./install.sh) which downloads, compiles the required software on your system. If you want to do it manually, look into download.sh to see the exact cvs checkout options which downloads the required asterisk and libpri versions. Yes, I know which libpri/asterisk versions bristuff downloads when using the included scripts (03/24/04). Problem is, I'd like to get the features / bugfixes from later versions. I'd especially like to try current oh323 drives, which require cvs head and don't compile against the versions usd by bristuff 0.2.2. Junghans has promised an update of the software. This was coming 'real soon' (Like when I say - 'I'll be there in 5 minutes' to the wife). I suspect it is even sooner now (promises of this last weekend) - so - sometime soon - and it should work against the current CVS HEAD. -- . . ___. .__ 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] help needed with read()
On Wed, 2004-06-23 at 17:12, Sathya wrote:
Hi,
Greatly appreciate if some one help me with the application read().
I have added a feature to reload asterisk from a phone...
it uses 'read' to get a 3 digit password
I was using '#' to end the sequence until I realised I could specify the
number should be only three digits long...
My voice prompts (posix-...) are described in the text comments...
; 307 = Restart Asterisk
exten = 307,1,DigitTimeout(4) ; Set Digit Timeout 4 seconds
exten = 307,2,ResponseTimeout(5); Set Response Timeout 5 sec
exten = 307,3,Read(Secret,posix-pass-restart-ast,3)
; to restart type the passwd
exten = 307,4,NoOp(${Secret})
exten = 307,5,Gotoif($[${Secret} = 123]?6:9)
exten = 307,6,Playback(posix-restarting) ; Restarting asterisk
exten = 307,7,Wait(1)
exten = 307,8,System(/usr/sbin/asterisk -rx reload)
exten = 307,9,Hangup
--
. . ___. .__ Posix Systems - Sth Africa
/| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE
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RE: [Asterisk-Users] Outgoing DTMF when using BRI i4l (Eicon Diva) - problems
On Tue, 2004-06-15 at 17:44, Mark Elkins wrote: On Tue, 2004-06-15 at 11:43, Shaun Ewing wrote: This is an issues with DTMF clamping, you need to use chan_capi to get DTMF working correctly. That's the last thing I wanted to hear :-( The jist of this is that i4l does not allow outgoing DTMF ??? ie - its broken??? Has anyone got the combination of Grandstream (I think this is irrelevent) + ISDN BRI (Dumb Cards - all seem to cause the problem) + i4l + outgoing DTMF working at all? What Version of Asterisk? So far - people who I've asked say No -- . . ___. .__ 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] Outgoing DTMF when using BRI i4l (Eicon Diva) - problems
On Tue, 2004-06-15 at 11:43, Shaun Ewing wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Williams Sent: Tuesday, 15 June 2004 6:55 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Outgoing DTMF when using BRI i4l (Eicon Diva) - problems This is an issues with DTMF clamping, you need to use chan_capi to get DTMF working correctly. That's the last thing I wanted to hear :-( Apparently my ISDN card (Eicon Diva 2.02 as I mentioned) supports CAPI, but I've only been able to find Windows drivers for it. The jist of this is that i4l does not allow outgoing DTMF ??? ie - its broken??? or is this just with the EICON dumb BRI card(s) ??? ...and only CAPI for ISDN cards actually works as desired? (ie - with outgoing DTMF) (if I could only successfully get the recipe for compiling the CAPI drivers for the EICON DIVA Dumb (2.02/2.01) cards :) -- . . ___. .__ 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] 'background' problem
On Sat, 2004-06-12 at 17:47, Mark Elkins wrote: I have a 'day' and a 'night' mode. In the day mode, I play a 'background' message which is interruptable by the pushing of a DTMF key - ie - all is normal. Let me try again... If I mix background announcements with SayUnixTime - then my IVR menu system breaks - DTMF tones are not recognised. Is this a Bug? What is the work around? My example was... exten = s,7,Playback(posix-welcome-afterhours) ; Welcome to Posix; Systems After hours support, Our business hours are Monday ; to Friday, 8am to 5pm. The time is now exten = s,8,SayUnixTime(||AIMP); A:Day, I:Hours, M:Minutes, P:am/pm exten = s,9,Playback(posix-welcome-afterhours-try) -- . . ___. .__ 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] 'background' problem
I have a 'day' and a 'night' mode. In the day mode, I play a 'background' message which is interruptable by the pushing of a DTMF key - ie - all is normal. In night mode - I decided to get smarter... I play two backgrounds with a 'sayunixtime' in between and now DTMF does nothing - the menu times out to my 'lets get the operator then'... If I change the three commands to a single 'playback' - everything works as expected. Is this because 'sayunixtime' breaks things? Should I use something else instead of the first 'playback'? This is with a very recent version of Head CVS. Code: exten = s,7,Playback(posix-welcome-afterhours) ; Welcome to Posix; Systems After hours support, Our business hours are Monday ; to Friday, 8am to 5pm. The time is now exten = s,8,SayUnixTime(||AIMP); A:Day, I:Hours, M:Minutes, P:am/pm exten = s,9,Playback(posix-welcome-afterhours-try) ; Please dial 1 ; for support, ...Blah... or Stay on the line for an operator Suggestions? -- . . ___. .__ 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] Musical interruptions
Whilst on a call, I'm getting the following... -- Started music on hold, class 'default', on SIP/phone3-a7d5 -- Playing 'pbx-transfer' (language 'en') -- Unable to find extension '#' in context 'default' -- Playing 'pbx-invalid' (language 'en') ie - without anyone pushing keys - I hear the music on Hold - as does the calling party. Are we somehow managing to sound like the tone for a '#' My BT100 phone is set up for DTMF=info This appears to happens quite randomly. Suggestions? I'm also getting quite a few... May 12 19:51:52 WARNING[98311]: chan_sip.c:542 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 152 (Non-critical Request) May 12 19:51:58 WARNING[98311]: chan_sip.c:542 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 153 (Non-critical Request) May 12 19:52:03 WARNING[4866068]: rtp.c:414 ast_rtp_read: RTP Read error: Resource temporarily unavailable May 12 19:52:03 WARNING[4866068]: rtp.c:414 ast_rtp_read: RTP Read error: Resource temporarily unavailable .. but am putting that down to running this extension (SIP Phone) over multiple 802.11 segments - in a semi-hostile environment. (I'm not the only person using 802.11 - there may be channel clashes) -- . . ___. .__ 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] Explain cidinternalcontexts?
On Mon, 2004-05-10 at 14:53, Philipp von Klitzing wrote:
Hi there,
could anyone drop a short line on what cidinternalcontexts exactly does
in voicemail.conf? The Wiki explanation isn't sufficient - at least not
for me... :-
From my understanding..
I have defined my internal extentions under the context [extensions] in
the file 'extensions.conf' - ie...
[extensions]
exten = 201,1,Macro(stdexten,${PMARK},203)
exten = 202,1,Macro(stdexten,${Support3},203)
exten = 203,1,Macro(stdexten,${Admin2},203)
(whatever)
If in voicemail.conf, I add the lines..
[general]
operator=yes; Allow '0 for an operator'
saycid=yes ; Speak the CLID on playback
; Define the internal contexts for the caller ID reporting function
; so that it says from extension for internal calls.
cidinternalcontexts=extensions
... then in a Voicemail from an extension, the message will be read out
Call from extension 203 - rather than someone that leaves a message
from outside that reads out Call from 18005556655
Also: How/where do I define an Operator extension?
After the normal Voicemail menu system (press 1 for Fred..), I have some
code that if the caller does not push any buttons (they are DTMF
challenged) - they get put through to my 'operators' phone. I added the
Operator function there.
exten = o,1,Goto(t,1) ; Someone dialed '0' for an operator?
;This is where the user is 'DTMF challenged', so ring the Switchboard
exten = t,1,Playback(posix-tryswitchboard) ;Lets try the switchboard
exten = t,2,Macro(stdexten,${Admin2},203) ; ..is my switchboard
exten = t,3,Hangup()
Cheers, Philipp
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Re: [Asterisk-Users] Problems when upgraded
On Mon, 2004-05-10 at 05:54, Simon Brown wrote: I have just installed one of the new TDM400 cards with an FXS and an FXO module into my * server. I also checked out the latest cvs head. I am using 7940 phones. Now I have some strange problems: 1. When in the VM menus, key presses do not register. 2. When I press hold on the 7940, it hangs up. Has anyone got any ideas? I'd like to think you've got the same problem as me - something in the new CVS head of the last day or so has stopped DTMF detection (eg - VM menus don't work). Since I reported DTMF Broken, there have been a few updates to CVS as well - maybe its fixed? - perhaps try a 'cvs update aterisk' and recompile - and let us know (I'm not sure its a sip.c problem - more a generic DTMF detection problem.) Use cvs checkout -r v1-0_stable asterisk and I expect everything will also work. -- . . ___. .__ 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] Problems when upgraded
On Mon, 2004-05-10 at 09:04, Mark Elkins wrote: On Mon, 2004-05-10 at 05:54, Simon Brown wrote: I have just installed one of the new TDM400 cards with an FXS and an FXO module into my * server. I also checked out the latest cvs head. I am using 7940 phones. Now I have some strange problems: 1. When in the VM menus, key presses do not register. 2. When I press hold on the 7940, it hangs up. Has anyone got any ideas? I'd like to think you've got the same problem as me - something in the new CVS head of the last day or so has stopped DTMF detection (eg - VM menus don't work). Since I reported DTMF Broken, there have been a few updates to CVS as well - maybe its fixed? - perhaps try a 'cvs update aterisk' and recompile - and let us know I tried my own suggestion - the CVS head has been fixed - DTMF no longer broken. Thanks Mark. -- . . ___. .__ 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] *** Asterisk sunday news: Read the sample configs, Luke!
On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote:
* Read the config sample files! (even if you're an Asterisk guru)
-
For those of you that have a working installation that you keep using, this is a
reminder to check into the configs/ directory of the Asterisk source tree, regardless
if you downloaded a tar ball or from CVS.
Good advice - so I do a CVS UPDATE... and 'say.c' is broken
gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT
-D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS
-DASTERISK_VERSION=\CVS-04/05/04-09:58:21\ -DINSTALL_PREFIX=\\
-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\
-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\
-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\
-DASTCONFPATH=\/etc/asterisk/asterisk.conf\
-DASTMODDIR=\/usr/lib/asterisk/modules\
-DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN
-DNEW_PRI_HANGUP-c -o say.o say.c
say.c: In function `ast_say_digit_str':
say.c:50: syntax error before '' token
say.c:57: warning: no return statement in function returning non-void
say.c: At top level:
say.c:58: syntax error before if
and in 'say.c' at about line 50
case ('#'):
snprintf(fn, sizeof(fn),
/digits/pound);
break;
default:
say.c
snprintf(fn, sizeof(fn), /digits/%c,
fn2[num]);
}
===
if((fn2[num] = '0') (fn2[num] =
'9')){ /* Must be in {0-9} */
snprintf(fn, sizeof(fn),
digits/%c, fn2[num]);
}
--
The lines that begin with say.c
-or is this just an error caused by CVS
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Re: [Asterisk-Users] *** Asterisk sunday news: Read the sample configs, Luke!
On Sun, 2004-05-09 at 14:33, Mark Elkins wrote: On Sun, 2004-05-09 at 09:59, Olle E. Johansson wrote: * Read the config sample files! (even if you're an Asterisk guru) - For those of you that have a working installation that you keep using, this is a reminder to check into the configs/ directory of the Asterisk source tree, regardless if you downloaded a tar ball or from CVS. Good advice - so I do a CVS UPDATE... and 'say.c' is broken ... The lines that begin with say.c Sorry folks... seems like a CVS Update did break - removed the file and re-updated. fine now. However - this could bit other people too.. in which case - delete the offending file - and update again (or always use 'cvs checkout' - less efficient - but..) -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
[Asterisk-Users] DTMF broken
Some CVS upgrade in the last day or two has broken the recognition of DTMF eg in Voicemail. I'm running the latest CVS as of now. I'm getting the error... *CLI -- Executing VoiceMailMain(SIP/phone1-e0dd, ) in new stack -- Playing 'vm-login' (language 'en') **Here I push a button** May 9 18:26:18 WARNING[98311]: chan_sip.c:5027 receive_info: Unable to retrieve DTMF signal from INFO message from [EMAIL PROTECTED] By re-installing an older (cvs checkout -r v1-0_stable asterisk) version - everything works fine again... thats with NO config changes at all.. Has someone removed some support for the transporting of DTMF (eg, info?) - I am using... dtmfmode=info in sip.conf with BudgeTone-100's (sent with absolutely no signatures or attachments) signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] DTMF broken
On Sun, 2004-05-09 at 20:48, Olle E. Johansson wrote: Mark, Could you please add a SIP debug message with the SIP INFO? I've done a debug with a working asterisk (V1.0) and the non-working asterisk. The trace is attached. :-)(debug - ascii text) When you say SIP INFO - what else are you asking for??? If its one of the 'sip show' commands - which one, and at what instance of time? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 This is a debug trace of Asterisk v1-0_stable - I'm dialing '310' which in extensions.conf looks like.. ; 310 = Access Voicemail - with full prompting exten = 310,1,VoicemailMain() I'm hanging up after 'dialing' 203 ... the 'bad' one follows after *CLI sip debug SIP Debugging Enabled *CLI Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2408 INVITE User-Agent: Grandstream BT100 1.0.4.63 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 345 v=0 o=phone1 8000 8000 IN IP4 160.124.48.121 s=SIP Call c=IN IP4 160.124.48.121 t=0 0 m=audio 5004 RTP/AVP 98 0 8 18 9 4 2 15 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:40 12 headers, 16 lines Using latest request as basis request Sending to 160.124.48.121 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format ULAW Found audio format GSM Found audio format UNKN Found description format iLBC Found description format PCMU Found description format PCMA Found description format G729 Found description format G722 Found description format G723 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 1309/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f To: sip:[EMAIL PROTECTED];user=phone;tag=as3564c06e Call-ID: [EMAIL PROTECTED] CSeq: 2408 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=6d4d7372 Content-Length: 0 to 160.124.48.121:5060 Sip read: ACK sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bK5550a5fc35b24bcb From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f To: sip:[EMAIL PROTECTED];user=phone;tag=as3564c06e Contact: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2408 ACK User-Agent: Grandstream BT100 1.0.4.63 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 11 headers, 0 lines Sip read: INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 160.124.48.121;branch=z9hG4bKf13dbe7ea5fc60a6 From: Phone One sip:[EMAIL PROTECTED];user=phone;tag=63f98f4e24e20f2f To: sip:[EMAIL PROTECTED];user=phone Contact: sip:[EMAIL PROTECTED];user=phone Proxy-Authorization: DIGEST username=phone1, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED] a;user=phone, nonce=6d4d7372, response=0142fb85eda2d7497992a0149d78e828 Call-ID: [EMAIL PROTECTED] CSeq: 2409 INVITE User-Agent: Grandstream BT100 1.0.4.63 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 345 v=0 o=phone1 8000 8000 IN IP4 160.124.48.121 s=SIP Call c=IN IP4 160.124.48.121 t=0 0 m=audio 5004 RTP/AVP 98 0 8 18 9 4 2 15 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:9 G722/8000 a=rtpmap:4 G723/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=ptime:40 13 headers, 16 lines Using latest request as basis request Sending to 160.124.48.121 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format ULAW Found audio format GSM Found audio format UNKN Found description format iLBC Found description format PCMU Found description format PCMA Found description format G729 Found description format G722 Found description format G723 Found description format G726-32 Found description format G728 Capabilities: us - 524302, them - 1309/0, combined - 12 Non-codec capabilities: us - 1, them - 0, combined - 0
Re: [Asterisk-Users] DTMF broken
On Sun, 2004-05-09 at 21:39, brian k. west wrote: What firmware you have on that BT101? And yes gnupg or what ever you use to sign your message did produce the attachemnt on this last one too. OK the gnuPG is off.. :-( Product Model:BT100 Software Version: Program--1.0.4.63 Bootloader--1.0.0.16 HTML--1.0.0.30 VOC--1.0.0.5 -- . . ___. .__ 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] Voicemail: upgraded?
I'm sure I saw a posting about someone updating the CVS with a more richly featured voicemail system. What happened? Am I wrong? Can't seem to find anything on this... -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] 729 licence on scsi
On Fri, 2004-05-07 at 23:00, Mark Spencer wrote: I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? If you'll just be patient for a little while, I'm working on new G.729 code which does NOT use the voiceage code and thus does NOT have their stupid SCSI problem. The new copy protection scheme will be based upon just the MAC address of your ethernet card, and WILL NOT DO ANYTHING WITH YOUR HARD DRIVE. **smootch** (I won't even ask you how long 'a little while' is either ;-) -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Transfering with Grandstream Phones
On Sat, 2004-05-08 at 20:43, Ryan Courtnage wrote: On 8-May-04, at 12:09 PM, Paul Tyreman wrote: I have a problem with my Grandstream phone. I have set it up to use DTMFMODE=info and I am able to transfer calls that have been made from that phone, but I am unable to transfer calls made TO that phone ?? I have the same problem (attempting to transfer a call made to my BT102 will result in that call being disconnected/hung). Workaround is to use '#' to transfer instead of the 'transfer' button on the phone. I also agree.. Using the '#' key is the only way to transfer. I'm running Software Version: 1.0.4.63 Nothing in the html menu mentions how 'transfer' might work - perhaps its a blank key waiting to be programmed one day??? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
[Asterisk-Users] H323 - Gatekeeper - asterisk - SIP config problems
After much reading and fiddling - I have the gnugk GateKeeper running and can make calls from the H323 phone to the sip phone. Voice works bi-directionally.. Calling from SIP to H323 gives me a problem... Both gnuGK and Asterisk are on the same box. Someone said this was OK. Others said No. I added a second IP (eth0:1) and told gnuGK that was HOME. How do I lock asterisk to the other (eth0) IP - then I think this might work or must I put gnuGK on a separate machine? ps Documentation on the combination of Asterisk, h323 and Gatekeepers is really well hidden - I ain't seen it anywhere. In oh323.conf - I have the section... [register] context=h323phone alias=Call from gwprefix=0 gwprefix=1 gwprefix=2 gwprefix=3 gwprefix=4 gwprefix=5 gwprefix=6 gwprefix=7 1 - [h323phone] in extensions.conf is identical to my [sip] section (for my internal phones) - seems to work OK. 2 - the 'Call from' appears now with the CLID on the displays of the H323 handsets - can't I get it to show the users name of the Extension? 3- the gwprefix lists then seems to make asterisk the default gateway for the numbers dialled that start with [0-7] - so asterisk completes the H323 handsets call - this seems ugly - and I have not seen anyone else's config doing anything similar. What dumb thing am I doing? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
[Asterisk-Users] 729 licence on scsi
I Purchased 4 licences for my SCSI only machine. I do have a CDROM - with a mounted CD. The Registration binary gives me a 'Segmentation Fault'. Is this like telling me I can't register the licence? Unfortunately - I only seriously scanned the mailing list after buying the keys Seems like the licence insists on using an IDE drive to create some sort of unique serial number.. Has anyone 'lost' their IDE and had problems? Who do I talk to now? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] 729 licence on scsi
On Fri, 2004-05-07 at 22:27, Billy Huddleston wrote: SO, do you have a IDE CDROM? Sorry - I should have said I do have an IDE CDROM - with a mounted CD (Yes) -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Extension Logic Question Help!! Park and Announce
On Wed, 2004-05-05 at 04:02, Kevin wrote:
I have an extension context that performs an assisted ParkandAnnounce
page. I create a temporary sound file to be played but I would like to
delete it after being used in the page park application. I cant figure
out how to delete the file after it is used in the context
ParkandAnnounce.
Can anyone offer a suggestion?
As this is the second time I've seen this - let me try.
I presume that the following is your current thoughts
exten = _7,1,Answer
exten = _7,2,Wait(1)
exten = _7,3,Playback(paging)
exten =
_7,4,Playback(/var/spool/asterisk/voicemail/default/${EXTEN:1}/greet
)
exten = _7,5,Playback(presspound)
exten = _7,6,Record(/tmp/pageperson%d:wav)
Change to: exten = _7,6,RECORD(/tmp/pageperson${EXTEN:1}:wav)
I have not seen anyone else use a printf '%d' construct anywhere else
- using the extension to be paged should be unique..
then - whereever you have 'RECORDED_FILE' - change it to ..
/tmp/pageperson${EXTEN:1} ??? I'd also kill the '^M'
exten = _7,7,Wait(1)
exten = _7,8,Playback(${RECORDED_FILE}})
exten = _7,9,Wait(1)
exten =
_7,10,ParkAndAnnounce(beep:beep:beep:/var/spool/asterisk/voicemail/d
efault/${EXTEN:1}/greet:${RECORDED_FILE}:hldonext:PARKED|60|Console/dsp|
extensions,${EXTEN:1},1) ^M
exten = _7,11,System(rm ${RECORDED_FILE})
Might change to System(/bin/rm /tmp/pageperson${EXTEN:1})
(full path name to 'rm')
exten = _7,12,Hangup
^
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RE: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Sun, 2004-05-02 at 22:07, Kevin Walsh wrote: Jon Lawrence [EMAIL PROTECTED] wrote: I emailed sales at digium asking whether the new module supported international (ie non bellcore) cli. The answer was yes, ... The Digium shop (http://store.yahoo.com/asteriskpbx/newitd1pofxo.html) says that I must have PCI 2.2 to make use of the card,... The BT CD50 and soldering iron plan is looking more and more like the one I'll be going with for now Um - Digium wants you to buy their hardware - but there is a CLID issue.. would it not make more financial sense to insert a dumb ISDN card (or two), and upgrade your PSTN to ISDN??? Would this not assist Digium in making sure CLID worked in the UK??? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
RE: Caller ID Re: [Asterisk-Users] Re: Support Digium
On Mon, 2004-05-03 at 00:11, David J Carter wrote: Mark J Elkins wrote Um - Digium wants you to buy their hardware - but there is a CLID issue.. would it not make more financial sense to insert a dumb ISDN card (or two), and upgrade your PSTN to ISDN??? Would this not assist Digium in making sure CLID worked in the UK??? Isn't this a bit like cutting of the nose to spite the face. UK PSTN lines costs £30 /Qtr UK ISDN costs £65 /qtr, you could buy two X100P's every year and still be in pocket by staying with PSTN. ISDN BRI is two lines - so that makes it £2.50 more per line - or £10 a year..?? no need to purchase the BT50 (a caller-ID unit? - at what cost? you need one per line? and an RS232 interface per unit?) There was a post on the list in the not to distant past where someone had written two small scripts for getting the information from a BT50 and a serial modification and passing it to asterisk. Still seems the best way in the interim. As has been said many times in the list Digium have given us this software, we don't have to give them a hard time in return. Not a fair payback. True - the software is excellent. If they sold an ISDN BRI 4-port card (like Fritz) - I'd buy it from them. No intentions of bad mouthing Digium... but USA != World -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Pattern matching rules for least cost routing
On Wed, 2004-04-21 at 01:03, Fran Boon wrote:
On Tue, 2004-04-20 at 23:21, Mark Elkins wrote:
No matter what is dialled - I always go out on the 'Default' line.
Swapping order makes no difference. If I comment out the 'default' - it
does match the 'Cell' pattern - and works.
Pattern-matching within a context is not done based on order at all.
include = cell
include = default
[cell]
exten = _00[78][234].,1,Playback(posix-cellphone)
exten = _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
[default]
exten = _0.,1,Playback(posix-defaultroute)
exten = _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
Thanks (to the three replies).
Ended up leaving the cell pattern matching where it was and putting just
the default [def-out] in its own context and 'including' that to the end
of the pattern matching with...
include= def-out
Little by little - I get to shape asterisk to the way I want it to
work..
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[Asterisk-Users] Make an H323 phone act like a SIP ohone
I have some Grandstream BT101 SIP phones. Work great (so far). I have some Planet VIP-101T H323 phones... how do I make them look/feel/act like a SIP phone I can dial to them from both Trunk + SIP's (ie - I've added 'oh323' libraries) What config do I add so that if I dial the * IP - they then at least act as an extension? Ideally I'd like to just pick up the handset, and dial a number - just like the SIP phones... Pointers please? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
[Asterisk-Users] Pattern matching rules for least cost routing
I've got two patterns I want to match on making an outgoing call...
(one day - to do Least Cost Routing for Cell/Mobile calls)
Firstly - I prefer '0' rather than '9' to get an outside line...
Either its a call to a mobile No... (072 -or- 082 -or- 083 -or- 084)
or its just another number to dial...
I added the following... the playback just advises me which 'route' is
being taken In 'extentions.conf' I have...
;Cell Phone call
exten = _00[78][234].,1,Playback(posix-cellphone)
exten = _00[78][234].,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
;Default catch all - just dial it
exten = _0.,1,Playback(posix-defaultroute)
exten = _0.,2,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}})
No matter what is dialled - I always go out on the 'Default' line.
Swapping order makes no difference. If I comment out the 'default' - it
does match the 'Cell' pattern - and works.
Shouldn't the number I dial match the longest match - not the shortest
match as it seems to be doing? Is there a way to change that logic??
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[Asterisk-Users] Speaking digits and time...
-- Executing DateTime(SIP/phone1-07ff, ) in new stack -- Playing '/var/lib/asterisk/sounds/digits/day-1' (language 'en') -- Playing '/var/lib/asterisk/sounds/digits/mon-3' (language 'en') -- Playing '/var/lib/asterisk/sounds/digits/h-19' (language 'en') This works - the pathname is complete - Joy. -- Executing SayDigits(SIP/phone1-0e7d, 203) in new stack -- Playing 'digits/2' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/3' (language 'en') This doesn't (silence). Path looks incomplete. Where in the source do I fix this -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] grandstream and stun
On Sun, 2004-04-18 at 11:26, Richard wrote: Hi, I noticed some issues with how grandstream handles stun test. GS is running version 1.0.4.50. The latest release of software for Grandstream (dunno if its the same for all phone??? - but for Product Model: BT100) is: Software Version: Program:1.0.4.55 Bootloader:1.0.0.14 HTML:1.0.0.24 One way to upgrade is set the phone's TFTP up to load from grandstream. (Line reads...) TFTP Server: 4.3.153.50 (for remote software upgrade and configuration) Then reboot. I had to reboot twice in order to update everything. I notice that the Grandstream phone tries to download from the TFTP server two interestingly names files... cfg000b82006e69 -and- cft.txt The first is cfg + Mac address of my phone. I'd guess this is could be my phone's config? Anyone know the format of the file - and how to make a phone dump its config? Anyone know the format of the cfg.txt file? Is there a definitive document on this anywere? (I've Searched Google) -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] SIP device rings once on busy before giving busy tone with dialplan
On Sat, 2004-04-17 at 15:58, Chris Orme wrote:
My dialplan is for the outgoing SIP call is:
exten = _00.,1,AbsoluteTimeout(3600)
exten = _00.,2,Dial(${TRUNK1}/${EXTEN:2},45,r)
exten = _00.,3,Answer
exten = _00.,4,Hangup
exten = _00.,103,Dial(${TRUNK2}/${EXTEN:2},45,r)
exten = _00.,104,Answer
exten = _00.,105,Hangup
(if call can go through on TRUNK1 send it out, if TRUNK1 is out of
capacity and therefore busy then try trunk 2 before giving up) if that is
busy (therefore it is likely the number really is busy then grab the
caller and hang them up (and they then hear 'busy').
Um - I'm probably missing the point entirely - but why are your trunks
not in a group and why are you not then using the group to dial out on?
(not posted to Asterisk - just you)
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[Asterisk-Users] (Newbie) help please?
What I've got... Software: Linux: Slackware 9.1 Asterisk: out of CVS - so its new. isdn4k-utils: to test the ISDN Card Hardware: PII Pentium 400Mhz (Its a test of concept machine) with 320Kb RAM 1 x ISDN BRI Card - DIVA EICON (Installed + working) 2 x Grandstream (Barbie?) BT100 SIP Phones. What Works.. I can call from one phone to the other... get read voicemail... I can dial from a PSTN phone the BRI Number - and get the * demo messages Whats been read.. Lots.. Andy's Getting Started (www.automated.it/guidetoasterisk.htm) and lots from http://www.voip-info.org/wiki-Asterisk+ISDN4Linux and I've followed almost every link from www.asterisk.org... All examples seem to include Digiums hardware :-( I'm looking for clean, clear examples with a generic ISDN card - which is my trunk line, and the two SIP phones. The numbering plan in South Africa is pretty simple 7 digits for local calls 12 digits for long distance Anyone in S.A. got some example configs to share with? Currently - I'm stuck with the message.. -- Executing Dial(SIP/phone1-082a, Modem/g1/8070590) in new stack Apr 17 00:09:00 WARNING[507919]: chan_modem.c:181 modem_call: Destination g1/8070590 requres a real destination (device:destination) -- Couldn't call g1/8070590 -- Hungup 'Modem[i4l]/ttyI1' ... when I dial '98070590' (9 for outside - which I'll make '0' one day!) (its late, head hurts, wife is loosing patience) help? hints? -- . . ___. .__ Posix Systems - Sth Africa /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 signature.asc Description: This is a digitally signed message part
