[Asterisk-Users] *@home 1.0 FWD inbound problems, 2 calls generated
Hi ALL Have installed [EMAIL PROTECTED] 1.0 On FWD DID's, appears that 2 calls are generated to the inbound extention. I have confirmed this on a number of friends boxes also. Does anyone have a fix for this ? I set the DID simply to a custom context and it did the same... Anyone have a way to fix this ? Here is the output.. -- Accepting AUTHENTICATED call from 65.39.205.121, requested format = 4, actual format = 4 -- Executing Goto(IAX2/[EMAIL PROTECTED]/5, ext-local|7020|1) in new stack -- Goto (ext-local,7020,1) -- Executing Macro(IAX2/[EMAIL PROTECTED]/5, exten-vm|[EMAIL PROTECTED]|7020) in new stack -- Executing SetVar(IAX2/[EMAIL PROTECTED]/5, FROMCONTEXT=exten-vm) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/5, 0?novm|1:3) in new stack -- Goto (macro-exten-vm,s,3) -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/5, 0?novm|1) in new stack -- Executing Macro(IAX2/[EMAIL PROTECTED]/5, dial|15|tr|7020) in new stack -- Executing AGI(IAX2/[EMAIL PROTECTED]/5, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- Accepting AUTHENTICATED call from 65.39.205.121, requested format = 4, actual format = 4 -- Executing Goto(IAX2/[EMAIL PROTECTED]/6, ext-local|7020|1) in new stack -- Goto (ext-local,7020,1) -- Executing Macro(IAX2/[EMAIL PROTECTED]/6, exten-vm|[EMAIL PROTECTED]|7020) in new stack -- Executing SetVar(IAX2/[EMAIL PROTECTED]/6, FROMCONTEXT=exten-vm) in new stack -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/6, 0?novm|1:3) in new stack -- Goto (macro-exten-vm,s,3) -- Executing GotoIf(IAX2/[EMAIL PROTECTED]/6, 0?novm|1) in new stack -- Executing Macro(IAX2/[EMAIL PROTECTED]/6, dial|15|tr|7020) in new stack -- Executing AGI(IAX2/[EMAIL PROTECTED]/6, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: request = dialparties.agi -- dialparties.agi: priority = 1 -- dialparties.agi: extension = s -- dialparties.agi: language = en -- dialparties.agi: accountcode = -- dialparties.agi: uniqueid = 1116763505.28 -- dialparties.agi: channel = IAX2/[EMAIL PROTECTED]/5 -- dialparties.agi: callerid = 0409839735 -- dialparties.agi: context = macro-dial -- dialparties.agi: type = IAX2 -- dialparties.agi: rdnis = unknown -- dialparties.agi: enhanced = 0.0 -- dialparties.agi: dnid = unknown dialparties.agi: Caller ID name and number are '0409839735' -- dialparties.agi: Added extension 7020 to extension map -- dialparties.agi: request = dialparties.agi -- dialparties.agi: Extension 7020 cf is disabled -- dialparties.agi: Extension 7020 do not disturb is disabled -- dialparties.agi: priority = 1 -- dialparties.agi: extension = s -- dialparties.agi: language = en -- dialparties.agi: accountcode = -- dialparties.agi: uniqueid = 1116763505.29 -- dialparties.agi: channel = IAX2/[EMAIL PROTECTED]/6 -- dialparties.agi: callerid = 0409839735 -- dialparties.agi: context = macro-dial -- dialparties.agi: type = IAX2 -- dialparties.agi: rdnis = unknown -- dialparties.agi: enhanced = 0.0 -- dialparties.agi: dnid = unknown dialparties.agi: Caller ID name and number are '0409839735' -- dialparties.agi: Added extension 7020 to extension map -- dialparties.agi: Extension 7020 cf is disabled -- dialparties.agi: Extension 7020 do not disturb is disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 7020 has call waiting disabled == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 7020 has call waiting disabled -- dialparties.agi: DbSet CALLTRACE/7020 to 0409839735 dialparties.agi: Dial string is SIP/7020|15|tr -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(IAX2/[EMAIL PROTECTED]/6, SIP/7020|15|tr) in new stack -- Called 7020 -- dialparties.agi: DbSet CALLTRACE/7020 to 0409839735 dialparties.agi: Dial string is SIP/7020|15|tr -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(IAX2/[EMAIL PROTECTED]/5, SIP/7020|15|tr) in new stack -- Called 7020 -- SIP/7020-22d9 is ringing -- SIP/7020-4abd is ringing -- Nobody picked up in 15000 ms -- Executing Wait(IAX2/[EMAIL PROTECTED]/6, 1) in new stack -- Nobody picked up in 15000 ms -- Executing Wait(IAX2/[EMAIL PROTECTED]/5, 1) in new stack == Spawn extension (macro-exten-vm, s, 5) exited non-zero on 'IAX2/ [EMAIL PROTECTED]/6' in macro 'exten-vm' == Spawn extension
[Asterisk-Users] Sipura 3000 FXO with Asterisk
Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work? The comments about it being an ugly hack arent really correct. The Sipura is really built for standalone useage wiht a sip provider however it does work well with asterisk. Follow this thread http://voxilla.com/forum-viewtopic-t-1335.html it works and it works **VERY** well :-) Pete ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk locking up - 99.9% CPU
Hello We are running Asterisk CVS 22/12/04 and pwlib/oh323 pandora version to work with our call agent. Unfortunately **VERY** frequently, asterisk stops responding and goes to 99.9% CPU. There is no debug output or other information that indicates there is a problem... Rather than continually restarting, can anyone make suggestions as to how we can track this down **OR** has anyone got the latest oh323/pwlb to work with CVS Head ? I see there is documentaiton on http://www.inaccessnetworks.com for the latest HEAD working with oh323 and pwlib... Any pointers would be appreciated -- Open WebMail Project (http://openwebmail.org) ___ 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] Calling Card Application - which one ?
Hello I'm interested in setting up a calling card application on asterisk. I noticed a number in the wiki, both free and commercial. To experiment with, I'm after a GNU licenced app...Which one would you recommend ? Regards..Peter -- Open WebMail Project (http://openwebmail.org) ___ 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] Dock-n-talk connection to asterisk
Hi Mike FOr a home solln, $1000 isnt overly cost effective and the payback period for myself would be too long. The technology is interesting and as everyone says, its **supposed** to work but I hate being the guinea pig. The dock-n-talk is interesting and supposedly it does work ok, the signalling for hangup is poor and not sure how that will work. I've got a few more irons in the fire with alternatives so I'll post my findings here as I go. Regards..Pete Message: 14 Date: Mon, 7 Mar 2005 23:32:36 +1100 From: Mike Sander [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dock-n-talk connection to asterisk Hi Peter. Look in last weeks (1/3/05) Sydney Morning Herald Tuesday IT liftout. They talk there about GSM gateways. It was made by Ericson I think, for around $1000. It's not meant for computer, rather as a FXO/FXS gateway to plug your house phone in for exactly the purpose you are talking about. Of course, if it is a FXO gateway, I'm sure a RJ cable (possibly crossover) will plug it in to a TD400 Digium card nicely to get what you want. I'm interested to know your progress, I have a few clients also interested in Sydney. Cheers Mike -- Open WebMail Project (http://openwebmail.org) ___ 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] What combination of pwlib and openh323 are
Hi Mark Funny you should ask this question, I just spent yesterday integrating building asterisk with h323 support to connect to a Cisco call agent.I cant say if it will work for you but it compiles and loads nicely ! I will be testing this evening # cd /root # wget http://www.voxgratia.org/releases/pwlib-Pandora_release-src-tar.gz # wget http://www.voxgratia.org/releases/openh323-Pandora_release-src-tar.gz # tar zxvf pwlib-Pandora_release-src-tar.gz # tar zxvf openh323-Pandora_release-src-tar.gz # cd /root/pwlib # ./configure make opt make install # cd /root/openh323 # ./configure make opt make install # echo /root/pwlib/lib /etc/ld.so.conf # may not be reqd # echo /root/openh323/lib /etc/ld.so.conf# may not be reqd # /sbin/ldconfig That gets pwlib and openh323 installed. Are you going to install ztdummy ? You will need to do this if you dont have a digium card installed.. If you need more help, drop me a message # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login (when prompted for a password, enter 'anoncvs') # cvs -z4 checkout -D '2004-12-22' zaptel libpri asterisk That gets the cvs version of Asterisk that works :-)You want this **specific** version of asterisk # cd /usr/src/asterisk/channels/h323 # make # cd /usr/src/asterisk # make # make install That will build the h323 support for asterisk. If I can help any furthur, drop me a mail.. I spent most of yesterday sorting it and was rewarded at midnight with a loading asterisk :-) Regards..Pete Date: Tue, 8 Mar 2005 11:41:19 +0800 From: [EMAIL PROTECTED] Subject: [Asterisk-Users] What combination of pwlib and openh323 are required to get Asterisk-oh323 v0.7.1 to compile To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii Hi there I have Asterisk running beautifully on our test server. Over the past few days I have been tearing my hair out trying to compile various versions of asterisk-oh323 on various versions of pwlib and openh323. pwlib is now up to 1.8.3 and openh323 is now 1.15.2 stable. asterisk-oh323 is currently 0.7.1 I have tried these three with many errors. I have tried 0.7.1 with pwlib 1.5.2 and openh 1.12.2 with no luck. I have tried asterisk-oh323 1.5.10 with pwlib 1.5.2 and openh323 and I still get errors. From the mailing list I have gleaned that this version of asterisk-openh323 won't work with the latest asterisk anyway, yet the readme in asterisk-oh323 says to use this version with the aforementioned versions of pwlib and openh323. I can't find the versions of pwlib and openh323 recommended in the asterisk-oh323-0.7.1 readme. The pwlib and openh323 projects always build without error. Asterisk built without errors and most everythings else. I am running a very basic Fedora Core 2 installation. What I would like to know is what is the recommended known good combination to use of asterisk-oh323, pwlib and oh323. Once I have a combination that should work, I can then ask more intelligent questions on how to get it to build properly if I still have errors. Help greatly appreciated. Regards Mark Dutton ___ 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] Cisco 7960
Date: Sun, 06 Mar 2005 20:03:52 +0100 From: Thomas Trepper [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco 7960 To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed Hi all, i am new to this list and i dot not know, if anybody had already the same problem. I have two cisco 7960 which i want to upgrade to sip. Has somebody already taken the upgrade-process for special hints and suggestions? I have already visited the cisco-page and i have read the proposal for the migration. Is there a special order of firmware- upgrades? Thanks a lot Thomas Thomas The asterisk-wiki is the best place to start. It will tell you that it is a 3 stage process if your currently on call-manager. You will need to load the version 3, then 5 and then 7 SIP firmware. I tried to load the version 7 straight away and of course it wouldnt work. Please read the wiki and all will be revealed. Dont expect very much from the cisco website at all ! Regards..pete ___ 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] Dock-n-talk connection to asterisk
Hi ALL I'm looking for feedback on how well this unit integrates into asterisk via an ata. Is the audio quality any good as thats the first thing to upset the wife if its no good. I'm looking for a reasonably priced GSM gateway 1800mhz for use in Australia that works with an ata. Quite happy to import something that works well... Currently PSTN to mobile is $0.40c per minute and going to a selected provider, it will only cost $0.05c per minute so the savings are enormous for me, hence my interest in the DOck-n-Talk Any feedback would be very much appreciated ! -- Open WebMail Project (http://openwebmail.org) ___ 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] FRS / FRS/GMRS 2-way radios as SIP clients
Hi Mark I've been involved with IRLP for about 5 years and am one of the original install team. I've gone through the emmotions of allowing other networks connect to IRLP and I know its caused some lots of headache. As far as a closed network goes, yes there is LOTS of passion to keep it HAM only and I'd have to support that notion. As far as non-radio users go, thats another issue. In the early days, I was WHOLLY against echo-irlp and it allowing headset users on the network. IRLP was designed to keep the RF in Amateur Radio. HOWEVER I see the benefits now and have certainly changed my views. IRLP **SHOULD** remain the domain of Amateurs **HOWEVER** asterisk has very cool functionality that could be used in IRLP... I run IRLP node 6000 and reflector 9500 and of course my own asterisk PBX and am now what I would consider a VOIP hobbyist.. Asterisk has given me renewed interest 73's Pete..vk2yx -- Open WebMail Project (http://openwebmail.org) -- Original Message --- From: Mark Phillips [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thu, 24 Feb 2005 17:43:36 -0500 Subject: Re: [Asterisk-Users] FRS / FRS/GMRS 2-way radios as SIP clients I got into SERIOUS trouble with the IRLP folks for trying to do this. They want a closed netowrk and won't entertain anything that could allow a non licenced ham from using their system. Mark Chris Albertson wrote: THere are a number of VOIP links used by ham radio. IRLP has to be the most popular. Then there is Echolink, Wires and some others. My plan was to wrte an Asterisk channel driver for each of these Asterisk then could provide inter-system bridging between the various ham VOIP networks, the PSTN and VOIP Telepony. --- Mark Phillips [EMAIL PROTECTED] wrote: Aha, I see where you're going with this. Firstly, why does it have to be SIP? Are you expecting to be able to have users pick up the phone and dial a radio? If not then there are loads of VOIP for radio apps out there. Many run under linux. All use sound card and serial port. Take a look at eqso.org They have a solution that is free and hooks you up to a load of other users using whatever radio you choose to use. Mark, KC2ENI Glenn Powers wrote: TC wrote: Any one know of software that allows 2-way radios as VoIP(SIP) clients, besides dingotel's usb mic cable trick ? http://www.dingotel.com/2way/requirements2way.asp They might be ok if the SIP client was not hardcode to their own SIP proxy Has anyone tried any hacks to get the 2-way radio SIP client to regsiter to a * box. hmm chan_frsgmfrs anyone? using the usb/mic cable under linux :) *The Asterisk http://www.asteriskpbx.org app_rpt project The integration of 2-way radio systems and reasonable telephony *http://www.zapatatelephony.org/app_rpt.html cheers, glenn ___ 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 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- ___ 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] 7960 Not Picking up new firmware.
I just upgraded a 7960 from Call Manager to Sip 7.3 Not having a clue, I couldnt load on 7.3 You will need to load on a version 3 sip image, then load on a verison 5 and then goto a version 7. As the doco says, its multistage Anyway, the 7960 works beautifully on the asterisk box, have the directory and services function going, I'm **MOST** impressed Regards..Pete -- Open WebMail Project (http://openwebmail.org) -- Original Message --- From: Ferguson, Michael [EMAIL PROTECTED] To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wed, 23 Feb 2005 11:43:46 -0500 Subject: RE: [Asterisk-Users] 7960 Not Picking up new firmware. Gary, Thanks again. You help has been invaluable. 'preciate it -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary G. Hendershot Sent: Wednesday, February 23, 2005 11:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 7960 Not Picking up new firmware. You have to change the image name in the OS79XX.txt and SIPDefault.cnf files to match the name of BIN file you are trying to load ... With versions of the firmware prior to 7.x, the name you put in the OS79XX.txt file and the SIPDefault.cnf files are the same; simply the BIN file name less the BIN extension ... As you get to version 7.x and up, the file name you put in OS79XX.txt is actually the name of a Universal Loader ... The name of the SIP binary image is entered in SIPDefault.cnf ... I got a help on this one from a pretty decent article on the WIKI at http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx ... Look at the section header Software Upgrade Requirements ... This gave me the clues I needed to get the 7.3 Sip image to load properly ... G.Hendershot -Original Message- From: Ferguson, Michael [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 23, 2005 10:43 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] 7960 Not Picking up new firmware. G'Day All. So I got the TFTP server all set up -thanks to much help from this list- the 7960 found it and updated to SIP the first firmware P0S30200. What I am now trying to do is upgrate through all the versions, as recommended, to the latest version, P003-07-3-00. I thought this would be accomplished by simply changing the sole line in the OS79XX.TXT file to P0S30203 and reboot the phone. But no success. Any pointers? Thanks Ferg ___ 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- ___ 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] Unable to create channel of type 'Zap' error
I'm trying to configure a 100xp fxo card for the first time but am not able to get the channel type ZAP recognised app_dial.c:743 dial_exec: Unable to create channel of type 'Zap' WHen starting asterisk with -vvvgc i see [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXS Kewlstart signalling -- Automatically generated pseudo channel == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) == Registered application 'CallingPres' == Manager registered action ZapTransfer == Manager registered action ZapHangup == Manager registered action ZapDialOffhook == Manager registered action ZapDNDon == Manager registered action ZapDNDoff == Manager registered action ZapShowChannels when I type zap show channel 1 i see: zap show channel 1 Channel: 1 File Descriptor: 17 Span: 1 Extension: Dialing: no Context: internal Caller ID string: Destroy: 0 InAlarm: 1 Signalling Type: FXS Kewlstart Owner: None Real: None Callwait: None Threeway: None Confno: -1 Propagated Conference: -1 Real in conference: 0 DSP: no Relax DTMF: no Dialing/CallwaitCAS: 0/0 Default law: ulaw Fax Handled: no Pulse phone: no Echo Cancellation: 128 taps, currently OFF Actual Confinfo: Num/0, Mode/0x Actual Confmute: No Actual Hookstate: Onhook *CLI The test I perform to dial is: exten = 47302410,1,Dial(Zap/1/47302410) I deal the number and get teh ZAP error. Can anoyone suggest soemthing I'm missing as I've never ued a ZAP interface before...and seems that http://www.voip-info.org is overloaded, not been able to get a page from them for a few days.. Any help would be aprpeciated ___ 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] giving up on x100p in Australia
Hi Manny I have a sipura 3000 connected to asterisk and I must say, its not a bad sollution. The units seem to be pre-configured to the US phone system and you need to do some work to get them working properly, namely the hangup tone detection... The audio levels aren't too bad, default they certainly need tweaking but it sounds ok. If you need to know anymore specifics, please contact me off list ! P.S a friend of mine at work received some new 100XP clone cards from the US and they solved his CLID and echo problems. So its all luck ! -- Open WebMail Project (http://openwebmail.org) -- Original Message --- From: Emanuele Venditti [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wed, 9 Feb 2005 14:33:19 +1100 Subject: [Asterisk-Users] giving up on x100p in Australia OK, I've spent way more time than I wanted to on getting an x100p clone to work in Australia. I'm happy to consider other (more functional) options. Does anyone have an opinion on both the Sipura 3000 and other Digium cards (like the TDM400P)? I need something that works with no much fuzz. I know the Sipura 3000 is cheaper the the TDM400P card. All I need is to channel my POTS line into Asterisk. Nothing else! And I only have one line! Any help would be appreaciated, Thanks, manny --- End of Original Message --- ___ 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] Digium X100P FXO Asterisk for Australia ?
Anyone in Australia using these cards ? I've been using a sipura and had to do a lot of research to get the thing to detect hangup and other subtilties on the Telstra network. These boards work OK in Australia ? Many Thanks Pete -- Open WebMail Project (http://openwebmail.org) ___ 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] Caller ID in AU
Nathan If you want more specific information for AUS, drop me a direct mail. My Sipura 3000 passes the PSTN call (on hook) to the asterisk box and also the CLIDNUM. My only problem is that the asterisk box then sends the caller-id to the handset connected to the sipura, I can get the username but the number never shows up even though I can see it in the asterisk messagesthats still soemthing I need to sort Pete -- Open WebMail Project (http://openwebmail.org) -- Original Message --- From: Nathan Alberti [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, 31 Jan 2005 09:54:13 -0500 Subject: Re: [Asterisk-Users] Caller ID in AU I have updated the Wiki with this info as I have seen it come up a few times. Nathan. Gary wrote: Don't forget Howard, that Caller-ID presentation is an extra chargeable service. has it been turned on on these lines and confirmed ?? (its handy to carry a caller-id in your kit for checking:-) On Sat, 29 Jan 2005 07:30:07 +1100, Howard Lowndes wrote: On Fri, 2005-01-28 at 19:02, Simon Brown wrote: Insert a Wait(2) before Answer OK, I'll try that. I have also done the suggested mod to the chan_zap.c module to make the default rings 2. Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes Sent: Friday, 28 January 2005 17:30 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Caller ID in AU Is anyone in AU successfully getting Caller ID from the analogue PSTN service? If so, what settings? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- ___ 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] Disa Syntax, some help please
Hi I'm confused by the asterisk WIKI syntax for DISA. I want to only let a CID of say 1234567 pass through DISA, which calls an extension of 333 In reading the documentation, I thought it should look like this exten = 333/1234567,1,Authenticate(1234567) exten = 333/1234567,2,DISA(no-password|my_context) This throws up all sorts of errors. I simply want the callerid to be tested, if its correct, the user should pass though DISA and onto my_context Any help here would be appreciated ! -- Open WebMail Project (http://openwebmail.org) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF on BT100
Set the grandstream to RFC2883 in your phone, this will work with asterisk. also define DTMFMODE=RFC2883 in sip.conf under the phone definition. Pete -- Open WebMail Project (http://openwebmail.org) -- Original Message --- From: Tomas Florian [EMAIL PROTECTED] To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Wed, 19 Jan 2005 15:27:44 -0700 Subject: RE: [Asterisk-Users] Asterisk not recognizing key beeps - DTMF on BT100 Thanks, this is what I found out so far: I have a Grandstream BT100, that is capable of doing both out of band and in band DTMF. But it doesn't work with either setting (I changed my sip.conf and the BT100 client phone accordingly of course) X-Lite works fine. I also upgraded the BT100 to have the newest firmware but that didn't help either. Are there some issues with BT100 phones and DTMF? Can I turn on DTMF debugging in asterisk somehow? Thanks, Tomas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yair Hakak Sent: Wednesday, January 19, 2005 2:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk not recognizing key beeps what endpoints are you using? You probably have a DTMF type mismatch between asterisk and your endpoint (IP phone or softphone) -yair On Wed, 19 Jan 2005 01:49:46 -0700, Tomas Florian [EMAIL PROTECTED] wrote: Hello, So far everything that I'm trying with asterisk is working except for this weird thing. When I try to call voicemail and it asks me for the password I enter it in but from the debug message I can see that it thinks I didn't enter anything in. Also when I'm leaving a message it sais press pound to end, but even if I press it 10 times it keeps on recording until I hang up. It just doesn't seem to recognize my key presses. I can dial, talk and do everything else ... but I just can't press keys during the call. I'm using a very simple setup from some quickstart with SIP and voicemail - nothing more than that. I remember that this used to work for me but then it stopped. I have no idea why, I couldn't find anything on the net about this problem. Any ideas? Thanks, Tomas ___ 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 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 mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Decode - this doesnt make sense :-)
Hello ALL This is bending my head and I'm hoping someone can help. Call flow as follows Sipphone - sipphone.com - asterisk - sipura 3000 (PSTN port) the sipphone is calling to my pstn line on the sipura. Works FANTASTIC, pin number on sipura entered, DTMF to PSTN decoded and number dialed, no problems ! IN the configurations below, the supra cant decode the DTMF, nor can the remote asterisk box decode the extension properly for the PSTN gateway mobile - ipkall.com - sipphone.com - asterisk - sipura 3000 mobile - sipura 3000 - asterisk - sipphone.com - asterisk - sipura sipphone - asterisk - sipphone.com - asterisk - sipura Using asterisk -c I see broken extension numbers ie: if it was extension 7010, it decodes 73, 0 701 etc With a straight sipphone in the top example, it works 100% reliable. If anyone could help I'd surely appreciate it as I'm out of ideas ! Regards..Pete -- Open WebMail Project (http://openwebmail.org) ___ 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 Decode - this doesnt make sense :-)
Richard Thanks for the response. All DTMF settings are RFC2833. The call through IPKALL I cannot control **their** DTMF settings so I'm a little confused as to what the settings should be...My friend and I have been carefull to use RFC2833. Any other suggestions ? :-) This is super weirdness.. Pete -- Open WebMail Project (http://openwebmail.org) -- Original Message --- From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tue, 18 Jan 2005 06:41:45 -0600 Subject: Re: [Asterisk-Users] DTMF Decode - this doesnt make sense :-) This is bending my head and I'm hoping someone can help. Call flow as follows Sipphone - sipphone.com - asterisk - sipura 3000 (PSTN port) the sipphone is calling to my pstn line on the sipura. Works FANTASTIC, pin number on sipura entered, DTMF to PSTN decoded and number dialed, no problems ! IN the configurations below, the supra cant decode the DTMF, nor can the remote asterisk box decode the extension properly for the PSTN gateway mobile - ipkall.com - sipphone.com - asterisk - sipura 3000 mobile - sipura 3000 - asterisk - sipphone.com - asterisk - sipura sipphone - asterisk - sipphone.com - asterisk - sipura Using asterisk -c I see broken extension numbers ie: if it was extension 7010, it decodes 73, 0 701 etc With a straight sipphone in the top example, it works 100% reliable. If anyone could help I'd surely appreciate it as I'm out of ideas ! Sounds like the various boxes are not defined to use the same dtmf signaling. Look at the various 'dtmfmode=' settings in asterisk, and the setting in the spa3k. They need to match. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- End of Original Message --- ___ 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
