[asterisk-users] Interact with IVR
I remember a thread similar to this a while ago but couldn't find. How do I make Asterisk to interact with an IVR? (Nothing fancy, just plain predictable voice menus like a conference bridge.) I get stuck at Dial(), which seems to wait for hangup after the other end picks up. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Command to disconnect a call
Uh maybe not... :) stop now will cause Asterisk to drop *all* calls and exit immediately. Kinda the equivalent of nuking a small city to kill one person. Paul Hales wrote: Stop now? PaulH On Sat, 2007-02-03 at 21:47 -0800, Yuan LIU wrote: Can I disconnect an arbitrary call using a console command? I remember reading something but can't find any more. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interact with IVR
Yuan LIU wrote: I remember a thread similar to this a while ago but couldn't find. How do I make Asterisk to interact with an IVR? (Nothing fancy, just plain predictable voice menus like a conference bridge.) I get stuck at Dial(), which seems to wait for hangup after the other end picks up. You can send dtmf to the IVR with the D option in the dial command. show application dial on the console will show you the syntax. Leo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400 stopped bridging outgoing FXO call
My TDM400/zaptel 1.2.10/Asterisk 1.2.13 suddenly stopped bridging outgoing FXO calls. If I make a call (from an FXS channel) to a PSTN destination, and the other side answers, Asterisk will show continued ringback on the FXS channel, while the PSTN side hears silence. No error message appears. If a call from PSTN terminates on the same FXO, Asterisk can still ring the FXS channel, and when FXS is picked up, normal conversation can complete. So the bridge is broken only when making outgoing calls. I am not sure what I have changed during the past few days. Any suggestions? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interact with IVR
From: Leo Ann Boon [EMAIL PROTECTED] Yuan LIU wrote: I remember a thread similar to this a while ago but couldn't find. How do I make Asterisk to interact with an IVR? (Nothing fancy, just plain predictable voice menus like a conference bridge.) I get stuck at Dial(), which seems to wait for hangup after the other end picks up. You can send dtmf to the IVR with the D option in the dial command. show application dial on the console will show you the syntax. Leo Thanks for the reply, Leo. D can do a maximum of one DTMF string. What next? (At least two levels for calling card and conference bridges, more for other things.) If people are using Asterisk as test equipment, there must be a way to conduct a dialogue with an IVR? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] TDM400 stopped bridging outgoing FXO call
From: Yuan LIU [EMAIL PROTECTED] My TDM400/zaptel 1.2.10/Asterisk 1.2.13 suddenly stopped bridging outgoing FXO calls. If I make a call (from an FXS channel) to a PSTN destination, and the other side answers, Asterisk will show continued ringback on the FXS channel, while the PSTN side hears silence. No error message appears. If a call from PSTN terminates on the same FXO, Asterisk can still ring the FXS channel, and when FXS is picked up, normal conversation can complete. So the bridge is broken only when making outgoing calls. It must be a hardware malfunction. I rebooted and the bridge comes back. There were no error messages - I turned up verbose and debug level to 8 and couldn't see anything. Yuan Liu I am not sure what I have changed during the past few days. Any suggestions? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400 noHangup
We have 2 line PBX office that connect to my VoIP Network that contain 2 Asterisk Server each server has FXO Digium. When I make call PBX-to-PBX via VoIP Network and when done my FXO port still active || || |PBX|-|Asterisk 1|-|-VoIP-||Asterisk 2|--|PBX| |Network| || || Is this TDM400 bug or there is something wrong with my zaptel configuration? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.2.14 and bristuff 0.2.0-RC8s
Hi All, How to install bristuff on asterisk 1.2.14? install scripts are trying to download and compile those versions: asterisk-1.0.10 zaptel-1.0.10 libpri-1.0.9 and I'm running: asterisk-1.2.14 zaptel-1.2.12 libpri-1.2.4 I only need Pickup application from bristuff to be able to pickup channel independent calls e.g. when I have incoming call from PSTN and I would like to answer ringing ext. from any SIP headset. Thank you in advance, Dominik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.14 and bristuff 0.2.0-RC8s
On Sun, Feb 04, 2007 at 11:24:36AM +0200, Dominik Zalewski wrote: Hi All, How to install bristuff on asterisk 1.2.14? i You don't. You install bristuff 0.3.0-PRE-1x . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and multicore processors
I'm specing out a new box to act as a tandem switch. It will have a TE410P with 4 x PRI and support IAX connections to four other boxes using predominantly ilbc and/or gsm. It also has 3 IAX trunks to Teliax for call routing also using gsm. No extensions actually terminate on the tandem, they're all switched to other boxes (highly distributed). On the PRI card, one goes to Embarq, the PSTN and two go to a legacy SX-200 which is being phased out. The fourth is a connection to an Adtran TSU-600 channel bank. Given this is a greenfield spec and we're building it from scratch, I'm looking at SuperMicro and their motherboards. Architecturally, I see the tandem as being CPU bound, if anything. Backbone is GigE connected to the server so I/O there isn't an issue and we aren't doing voicemail on it so it isn't diskbound. Primarily the load will in in transcoding between the PRI channels and the IAX channels. We're looking at probably no more than 100 calls simultaneously. All the remote boxes use the same codec on the channels, so it doesn't have to transcode for inter box comm's. How well does asterisk spread itself out over multiple CPU's (aka Cores). I'm looking at their 2xQuadCore (clovertown) motherboards and was spec'ing CPU's. I know this is a religious issue in some circles, but is it better to have one Quad core as fast as you can buy (4 CPU's) or 2 x Quad core at a lower speed (8 CPU's). Obviously, I've got to shoehorn a budget here and can do 8 for the price of 4 if * will spread itself out. If transcoding is threaded and doesn't deadlock for a single resource, it seems 8 cores would be better than 4. Thoughts? Thanks for any input. Eric ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help sought: Asterisk H.323, Cisco IOS Gatekeeper(s) intra-zone call routing and TETRA
All, I'm haveing a bit of trouble getting my head around H.323 and call routing with Gatekeepers, Zones and intra-zone calls - hopefully someone who is more informed in things H.323 will be able to point me in the right direction...? I already have a mature network of Asterisk boxes dotted around the UK and overseas with hundreds of extensions and our own number-plan/dial-plan in the form 8EE where 'EE' is a two digit exchange number and '' is the four digit extension number, so for example my desk phone (a Cisco 7970G as it happens) at home is 8202001 and other phones around the house 8202002, 8202003 etc. I have a Cisco 2621XM connection to the internet with NM-2V and some VIC-2EM cards and have my home Asterisk box hooked up to ham radio and other interesting audio sources which I use Asterisk ooh323 to make H.323 connections to the Cisco and its simple, reliable and just works. I'm working on an RD project with TETRA -- a digital trunked radio system -- the hardware of which comes from Damm Cellullar (www.damm.dk) and is called TetraFlex - the system supports point-to-point (half duplex), group comminication and PABX/PSTN interconnect via H.323. Further investigation shows the system controller to be built on Windows-XP embedded and use the OpenH323 stack to provide H.323 connectivity, however the use of a Gatekeeper is mandatory in their set up. So, I have the following: - a network of Asterisk boxes using 8EE numbering with IAX2 between sites, hundreds of SIP and SCCP phones - a TETRA system using 'mobiles' that I have numbered 817 to fit my number plan, a H.323 setup that requires a Gatekeeper - an Asterisk box (1.2.14 + asterisk-addons-1.2.5) on the same LAN segment as the TETRA system with OOH323 compiled up and working - the devices are as follows: 192.168.1.5is the Asterisk box 192.168.1.6is the Cisco 2621XM GateKeeper 192.168.1.7is the Damm TETRA system Where I am stuck is partly with the H.323 concepts and partly with the implementation... First I went in search of a H.323 Gatekeeper and tried to build the GNU GK from source on another Fedora Core 6 box which turned out to be a nightmare (missing libraries, the OpenH323 libraries appear to be deprecated in favour of Opal? conflicting versions of pwlib, problems with yacc/bison building the ASN.1 parser, etc. etc.) so next I turned to Cisco. I have a spare 2621XM router with c2600-jsx-mz.123-22.bin which includes a H.323 gatekeeper... great... so I've configured it up (all three lines of it :o) so far: ! gatekeeper zone local TETRA tubby.org 192.168.1.6 no shutdown ! and can get both Asterisk and the Damm TETRA equipment to register with the Gatekeeper, like this: router-h323-gw#show gatekeeper endpoints GATEKEEPER ENDPOINT REGISTRATION CallSignalAddr Port RASSignalAddr Port Zone Name TypeFlags --- - --- - - - 192.168.1.5 1720 192.168.1.5 13030 TETRA UNKN-GW H323-ID: PABX H323-ID: Asterisk E164-ID: 100 Voice Capacity Max.= Avail.= Current.= 0 192.168.1.7 1720 192.168.1.7 1085 TETRA UNKN-GW H323-ID: DAMM Voice Capacity Max.= Avail.= Current.= 0 Total number of active registrations = 2 router-h323-gw# Over at the Asterisk box I have the following in ooh323.conf: [general] ;Define the asetrisk server h323 endpoint bindaddr=0.0.0.0 port=1720 gateway=yes h323id=Asterisk e164=100 gatekeeper = 192.168.1.6 [cisco-gk] type=peer ip=192.168.1.6 port=1720 context=from-h323 disallow=all allow=alaw allow=ulaw allow=gsm rtptimeout=60 dtmfmode=rfc2833 h323id=PABX I have the Cisco document Understanding H.323 Gatekeepers available here: http://www.cisco.com/warp/public/788/voip/understand-gatekeepers.html While there is a diagram (top of page 10) in the Cisco document which appears to show exactly what I want to make there's scant information on how to configure it and hence its limited help with what I am trying to do, namely gateway between two different H.323 gateways in a single zone and route calls between the two gateways - for example: Route Where Description 817 TETRA to mobiles/portables on the Damm TETRA system 8.. PABXto numbers in the rest of my number plan 0. PABXthrough the PABX on to the PSTN 9.. PABXthrough the PABX to emergency 911/999 I can see how this be acheived if I had two Gatekeepers and two zones, say one called tetra and one called asterisk by using zone prefixes and zone remote to route between them and putting one of
[asterisk-users] SMDI support on trixbox
Last question for the day, I promise. On voip-info.org and trixbox.org, I found some old threads on MWI via SMDI. Has this been rolled into Tbox or has anyone successfully rolled it in after the fact. As part of our longterm plan, I'd like to move the legacy PBX to Tbox and pass MWI back to it via SMDI, like the current system. As we drop the extensions on the legacy side and move to all IP phones, their VM will stay with them then. Thanks EKG ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Talk without gmail accout?
Well, it's like trying to check your hotmail.com email account from netscape.net - it just isn't going to work. What you can do, however, is talk to people on google talk from other jabber systems, just like you can send a netscape.net user an email from hotmail.com - Ian Hailey [EMAIL PROTECTED] wrote: Hello all, I am having trouble getting gtalk to work with my account which is not using a gmail.com email address. When I do this there an error from the Jabber module: [Feb 3 20:51:17] ERROR[6286]: res_jabber.c:573 aji_act_hook: JABBER: Node Error [Feb 3 20:51:17] WARNING[6286]: res_jabber.c:1495 aji_recv_loop: JABBER: Got hook event. JABBER: gtalk_account INCOMING: stream:errorhost-unknown xmlns=urn:ietf:params:xml:ns:xmpp-streams/str:text xmlns:str=urn:ietf:params:xml:ns:xmpp-streamsSet the 'to' attribute of stream element to the domain part of the user's JID. Example: to='gmail.com'./str:text/stream:error/stream:stream So does this only work if you have email accounts from gmail.com? Thanks Ian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason Parker Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interact with IVR
Yuan LIU wrote: From: Leo Ann Boon [EMAIL PROTECTED] Yuan LIU wrote: I remember a thread similar to this a while ago but couldn't find. How do I make Asterisk to interact with an IVR? (Nothing fancy, just plain predictable voice menus like a conference bridge.) I get stuck at Dial(), which seems to wait for hangup after the other end picks up. You can send dtmf to the IVR with the D option in the dial command. show application dial on the console will show you the syntax. Leo Thanks for the reply, Leo. D can do a maximum of one DTMF string. What next? (At least two levels for calling card and conference bridges, more for other things.) If people are using Asterisk as test equipment, there must be a way to conduct a dialogue with an IVR? Use w in the D() string to wait for .5 second. Use multiple w's to wait longer. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detecting answer with an analogue card
Hello, I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I would like use analogue lines for outboud calls. How is it possibile to detect ANSWER? - answeronpolarityswitch does not seem to work in Italy - call progress does not give safe results, sometimes calls get billed, sometimes not Am I forced to buy an ISDN adapter or could there be a solution (maybe tweaking some configuration parameter...)? Thanks and best regards Stefano C. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.14 and bristuff 0.2.0-RC8s
On Sun, Feb 04, 2007 at 11:54:14AM +0200, Tzafrir Cohen wrote: On Sun, Feb 04, 2007 at 11:24:36AM +0200, Dominik Zalewski wrote: Hi All, How to install bristuff on asterisk 1.2.14? i You don't. You install bristuff 0.3.0-PRE-1x . BTW: consider also asking on http://lists.three-dimensional.net/mailman/listinfo/bristuff-users Other people interested in bristuff are encoureged to subscribe ;-) -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 noHangup
Mochamad Susantok wrote: We have 2 line PBX office that connect to my VoIP Network that contain 2 Asterisk Server each server has FXO Digium. When I make call PBX-to-PBX via VoIP Network and when done my FXO port still active || || |PBX|-|Asterisk 1|-|-VoIP-||Asterisk 2|--|PBX| |Network| || || Is this TDM400 bug or there is something wrong with my zaptel configuration? Thanks Your PBX is not signaling Asterisk that the line has disconnected. Analog FXS ports on PBX do not generally provide this kind of signaling. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bristuff mailinglist
Tzafrir Cohen already mentioned it in a reply to someone else: There's a bristuff-users mailinglist now. If you are interested in bristuff or are using it consider subscribing to it. The webinterface is here: http://lists.three-dimensional.net/mailman/listinfo/bristuff-users The list address is: [EMAIL PROTECTED] This mailserver is running greylisting, so you might see a temporary failure on the first message. No need to panic, your mailserver (or your ISP's mailserver) will send it again and then the server will accept it. Too bad we have to do this but we dont need spam right ;) Greetings, -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting answer with an analogue card
Hi, On Sun, 2007-02-04 at 16:17 +0100, Stefano Corsi wrote: Hello, I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I would like use analogue lines for outboud calls. How is it possibile to detect ANSWER? you cannot. it's analogue, no signalling is done on it. unless you write dsp routines to detect the right things at the right moment :) if you want exact cdr records, you must go digital. greetings, matteo. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting answer with an analogue card
I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I would like use analogue lines for outboud calls. How is it possibile to detect ANSWER? you cannot. it's analogue, no signalling is done on it. unless you write dsp routines to detect the right things at the right moment :) if you want exact cdr records, you must go digital. There's still something I don't understand: when using a simple modem on an analog line, you get correct answers from the modem: NO ANSWER, BUSY, NO DIALTONE, etc... why is this possible with these TDM2400 cards that cost twenty times as much? I know that I'm probably missing something... could you help me understand what? Thanks Stefano ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error dialing a SIP user. chan_sip.c:1994 create_addr: No such host
Indeed. The problem was the ).
thanks to all who helped me debug this...my eyes are not so young anymore...
On 2/3/07, jacobso1 [EMAIL PROTECTED] wrote:
hi,
i think the problem is here :
exten = _321[0123],n,Dial(SIP/${EXTEN}),30,to)
|
replace with
exten = _321[0123],n,Dial(SIP/${EXTEN},30,to)
note, i removed the parenthesis ')' after the {EXTEN}
this should do
regards,
jacobson
---
Scarlet ONE - Combine ADSL with unlimited fixed phone and save 400 euros
http://www.scarlet.be
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Re: [asterisk-users] bristuff mailinglist
That's a very good news !! Congratulations to Tzafrir for it ! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting answer with an analogue card
Stefano Corsi wrote: I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I would like use analogue lines for outboud calls. How is it possibile to detect ANSWER? you cannot. it's analogue, no signalling is done on it. unless you write dsp routines to detect the right things at the right moment :) if you want exact cdr records, you must go digital. There's still something I don't understand: when using a simple modem on an analog line, you get correct answers from the modem: NO ANSWER, BUSY, NO DIALTONE, etc... why is this possible with these TDM2400 cards that cost twenty times as much? I know that I'm probably missing something... could you help me understand what? Because the far end sends a carrier tone. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zap FXS slow to reset?
I have the following dialplan (segment) that isn't working as I expected it to: exten = s,n,Dial(Zap/1SIP/202SIP/203,18) exten = s,n,Dial(Zap/1SIP/201SIP/202SIP/203,42) The plan was to have SIP/201 added to the group of ringing phones after 3 or so rings. What ends up happening, though, is the Zap/1 phone STOPs ringing when the dialplan falls through to the second line. The log reports: Feb 4 09:20:52 WARNING[26889] chan_zap.c: Unable to ring phone: Device or resou rce busy So, it seems to me that the Zap interface isn't ready yet to take another call. Is there another way I can accomplish the same thing? This method seems to work great when I'm dealing with all SIP phones. This one kind of blindsided me. Thanks. ttyl srw ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dnsmgr died?
Hi, I turned on dnsmgr in 1.2 and it worked for a few weeks. Suddenly, no iax2 providers were working. All of them were unreachable. My own fixed ip phones were. I disabled the dnsmgr and now the IAX providers are working again. No big deal, but it's odd that this happened this way. Anyone else had problems with it? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Interact with IVR
Yuan LIU wrote: From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] I remember a thread similar to this a while ago but couldn't find. How do I make Asterisk to interact with an IVR? (Nothing fancy, just plain predictable voice menus like a conference bridge.) I get stuck at Dial(), which seems to wait for hangup after the other end picks up. You can send dtmf to the IVR with the D option in the dial command. show application dial on the console will show you the syntax. Leo Thanks for the reply, Leo. D can do a maximum of one DTMF string. What next? (At least two levels for calling card and conference bridges, more for other things.) If people are using Asterisk as test equipment, there must be a way to conduct a dialogue with an IVR? Use w in the D() string to wait for .5 second. Use multiple w's to wait longer. That's a neat trick. Thanks, Eric. To expand on the concept of a dialogue, how sophisticated can it get? I'm thinking of less predictable IVR's. Although I'd not think of Asterisk as a replacement for Hammer, I'm imagining one Asterisk doing functional test of another. Here's the scenario: *A (tester) dials *B (testee), invoking an IVR. Each menu item in *B ends (or starts, if background) with a specific DTMF string that *A can intercept so *A can easily identify where in the tree it is. This way, I can program *B to walk every path in the IVR. Is this possible? Not in the dialplan with Dial(). You MIGHT be able to do so with a .call file and an AGI script, but I don't know for sure. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting answer with an analogue card
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED] Stefano Corsi wrote: I have two TDM2400 card with some 40 FXS modules and 4 FXO modules. I would like use analogue lines for outboud calls. How is it possibile to detect ANSWER? you cannot. it's analogue, no signalling is done on it. unless you write dsp routines to detect the right things at the right moment :) if you want exact cdr records, you must go digital. There's still something I don't understand: when using a simple modem on an analog line, you get correct answers from the modem: NO ANSWER, BUSY, NO DIALTONE, etc... why is this possible with these TDM2400 cards that cost twenty times as much? I know that I'm probably missing something... could you help me understand what? Because the far end sends a carrier tone. Makes me wanting Asterisk to be able to drive MODEM protocols. Since X100P is just a soft MODEM, how possible is this? (If not V.92, V.22 would be useful.) Suppose Intel (or Motorola) has a Linux driver for the card, can it be used in conjunction with Asterisk so I can use AGI to do the MODEM part? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400 noHangup
What PBX do you have connected to the Asterisk servers? My experience is that most PBXs do not provide a disconnect signal on their analog station ports. I have had the most success with disconnects on Avaya PBXs. Nortel analog stations last I tested with any, did not provide a disconnect signal. On these other PBXs you will need to use band-aids like having zaptel monitor for busy or dial tone when the analog station hangs up. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Zap FXS slow to reset?
From: Scott Walde [EMAIL PROTECTED] I have the following dialplan (segment) that isn't working as I expected it to: exten = s,n,Dial(Zap/1SIP/202SIP/203,18) exten = s,n,Dial(Zap/1SIP/201SIP/202SIP/203,42) Interestingly, although the Asterisk Manual (by Mark Spencer and so on) contains an almost identical sample plan fragment, in reality, it seems to need a Wait() in between to reset the Zap channel in the first Dial(). I tested with TDM400. Inserted a 1-2 sec wait and it did what you wanted. Yuan Liu The plan was to have SIP/201 added to the group of ringing phones after 3 or so rings. What ends up happening, though, is the Zap/1 phone STOPs ringing when the dialplan falls through to the second line. The log reports: Feb 4 09:20:52 WARNING[26889] chan_zap.c: Unable to ring phone: Device or resou rce busy So, it seems to me that the Zap interface isn't ready yet to take another call. Is there another way I can accomplish the same thing? This method seems to work great when I'm dealing with all SIP phones. This one kind of blindsided me. Thanks. ttyl srw ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem loading AstDB into variable on restart
I define
[globals]
myvar = ${DB(store/myvar)}
---
But when I want to use ${myvar} in the dial plan, I found that the variable
is null when Asterisk is restarted. Only a reload would force globals to
read AstDB. Other variables in globals loads fine.
Any idea? (Asterisk 1.2.13)
Yuan Liu
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[asterisk-users] Continue line in config files?
Is there anything that allows a logical line to extend to the next physical line? Printed files are so hard to read with blind line wraps - and my printer doesn't even automatically wrap. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue line in config files?
how would a line be soo loogg? On 2/4/07, Yuan LIU [EMAIL PROTECTED] wrote: Is there anything that allows a logical line to extend to the next physical line? Printed files are so hard to read with blind line wraps - and my printer doesn't even automatically wrap. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] musiconhold restarts for every extension
On Fri, 2 Feb 2007 17:56:26 -0500 Wes Baehr [EMAIL PROTECTED] wrote: The problem can be reproduced in the same way by putting a caller on hold, unholding, and holding again. The MOH restarts from the beginning of whichever file it was playing last. (I have random enabled, so it randomly picks a please wait for the next blah blah blah file). (I'm using 1.4 release). Does this occur for you as well? yes ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP privacy headers
Hi, Out ITSP has told us to user SIP privacy headers to hide outbound caller ID. Does anyone know how or if this can be done in Asterisk. I tried exten = s,3,SIPAddHeader(privacy=on) prior to executing Dial but no luck. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP privacy headers
Look here: http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party- ID+header From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop Sent: Sunday, February 04, 2007 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP privacy headers Hi, Out ITSP has told us to user SIP privacy headers to hide outbound caller ID. Does anyone know how or if this can be done in Asterisk. I tried exten = s,3,SIPAddHeader(privacy=on) prior to executing Dial but no luck. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
Im glad to let you know that finally I invested some time to make work Unicall in Asterisk 1.4, I must say not much testing could be done since I have no hardware available ( cards, servers ), however a friend was able to test it with a couple of calls with success, I need you to test this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/ Kind Regards Moises Silva -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP privacy headers
thanks for that. Do you know what P-Asserted-Identity needs to be set to to hide caller ID via privacy headers? On 2/5/07, Darryl Dunkin [EMAIL PROTECTED] wrote: Look here: http://www.voip-info.org/wiki/view/P-Asserted-Identity+and+Remote-Party-ID+header -- *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Eric Bishop *Sent:* Sunday, February 04, 2007 15:43 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] SIP privacy headers Hi, Out ITSP has told us to user SIP privacy headers to hide outbound caller ID. Does anyone know how or if this can be done in Asterisk. I tried exten = s,3,SIPAddHeader(privacy=on) prior to executing Dial but no luck. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap FXS slow to reset?
Yuan LIU wrote: exten = s,n,Dial(Zap/1SIP/202SIP/203,18) exten = s,n,Dial(Zap/1SIP/201SIP/202SIP/203,42) Interestingly, although the Asterisk Manual (by Mark Spencer and so on) contains an almost identical sample plan fragment, in reality, it seems to need a Wait() in between to reset the Zap channel in the first Dial(). I tested with TDM400. Inserted a 1-2 sec wait and it did what you wanted. I thought about doing that but was worried about the condition where someone answers a phone during that 1 second (which is actually quite likely to happen) and they only get dialtone rather than answering the incoming call. Has this not been an issue for you? ttyl srw ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do I debug?
I had my setup working properly under 1.2 and after a disk crash I decided that I wanted to try Asterisk 1.4. So far I can transfer between phones and I can dial out. What I can't get working is to get an SPA-3102 to 'send the calls' to Asterisk. I have the device added to the sip.conf file and it shows up in users and peers (but not in registry). Where do I start with debugging? -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://www.linuxha.com/ Main site http://linuxha.blogspot.com/My HA Blog Author of: Linux Smart Homes For Dummies ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FreeBSD Compile Errors
Hi everyone: I'm trying to compile Asterisk on FreeBSD 6.0-RELEASE and I'm getting the following error: cc -O2 -fno-strict-aliasing -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -DMAKE_VALGRIND_HAPPY -I/usr/local/include -L/usr/local/lib -I/usr/local/include/spandsp -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -I/usr/local/include -L/usr/local/lib -fPIC -c -o app_page.o app_page.c cc -shared -Xlinker -x -o app_page.so app_page.o gmake[1]: *** No rule to make target `app_rxfax.o', needed by `app_rxfax.so'. Stop. gmake[1]: Leaving directory `/usr/ports/net/asterisk/work/asterisk-1.2.13/apps' gmake: *** [subdirs] Error 1 *** Error code 2 Anyone have any idea what's going on? I just did a CVSUP so I know I'm running the latest verion. I CSV'd to: *default release=cvs tag=RELENG_6_0. Any thoughts??? -Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TDM400 noHangup
I am don't sure what PBX we have, but when i hangup i hear like tut tut tut in the other side, Is that not disconnected signal ? FYI when i take call from PBX to VoIP client or vice versa. it's ok. What do you think, are there is have some relation with my problem ? What PBX do you have connected to the Asterisk servers? My experience is that most PBXs do not provide a disconnect signal on their analog station ports. I have had the most success with disconnects on Avaya PBXs. Nortel analog stations last I tested with any, did not provide a disconnect signal. On these other PBXs you will need to use band-aids like having zaptel monitor for busy or dial tone when the analog station hangs up. Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: why there havn'tapp_meetme.sofileaboutasterisk1.4.0?
This would do it, but a better way would be to specify --with-zaptel=PATH (PATH is the directory of zaptel sources) when running configure. If you already did a build you probably want to run make dist-clean before running configure again. Best regards, Alex Alex Epshteyn Third Lane Technologies, LLC http://www.thirdlane.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bill Gibbs Sent: Thursday, February 01, 2007 6:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Re: why there havn'tapp_meetme.sofileaboutasterisk1.4.0? Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of ?? Sent: Thursday, February 01, 2007 9:01 AM To: Asterisk Users Mailing List - No Subject: Re: [asterisk-users] Re: why there havn't app_meetme.sofileaboutasterisk1.4.0? Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. You have to compile and install Zaptel first, for asterisk to build meetme. -- -- Steven http://www.glimasoutheast.org 李君 [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-02-01 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels ( when I use asyncgoto)
Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto(SIP/111-086497c8, SIP/113-08674628|dynamic-nway|111|1) in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting async goto (SIP/113-08674628) to dynamic-nway,111,1 Feb 2 16:53:10 DEBUG[4218]: channel.c:2834 ast_channel_masquerade: Planning to masquerade channel SIP/113-08674628 into the structure of AsyncGoto/SIP/113-08674628 Feb 2 16:53:10 DEBUG[4218]: channel.c:2847 ast_channel_masquerade: Done planning to masquerade channel SIP/113-08674628 into the structure of AsyncGoto/SIP/113-08674628 Feb 2 16:53:10 DEBUG[4218]: channel.c:2972 ast_do_masquerade: Got clone lock for masquerade on 'SIP/113-08674628' at 0x8677314 Feb 2 16:53:10 DEBUG[4218]: channel.c:3154 ast_do_masquerade: Putting channel SIP/113-08674628 in 2/2 formats Feb 2 16:53:10 DEBUG[4218]: channel.c:3189 ast_do_masquerade: Released clone lock on 'AsyncGoto/SIP/113-08674628ZOMBIE' Feb 2 16:53:10 DEBUG[4218]: channel.c:3198 ast_do_masquerade: Done Masquerading SIP/113-08674628 (6) Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/111-086497c8 and AsyncGoto/SIP/113-08674628ZOMBIE Feb 2 16:53:10 DEBUG[4218]: app_dial.c:1636 dial_exec_full: Exiting with DIALSTATUS=ANSWER. -- Executing Set(SIP/111-086497c8, DYNAMIC_FEATURES=) in new stack -- Executing Goto(SIP/111-086497c8, dynamic-nway|111|1) in new stack -- Goto (dynamic-nway,111,1) == Channel 'SIP/111-086497c8' jumping out of macro 'nway-start' Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/112-08641920 and SIP/111-086497c8 I want to know why there are this warning? How can I fix it? With Regards, Amy 李君 [EMAIL PROTECTED] 2007-02-02 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk cann't redirect the calling party to anothere Exten.
Hi All, I use the Asterisk Manager Interface to redirect the channels. There have two channels : SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456) SIP/612-5456 [EMAIL PROTECTED]:10 Up Dial(SIP/[EMAIL PROTECTED] Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel: SIP/voip_out_22-809c Exten: 111 Context: meetme-test Priority: 1 Then , the channel named SIP/voip_out_22-809c has been transfered to the conference 111. But, the channel named SIP/612-5456 has been hangup automatic. The context meetme-test is : [meetme-test] exten = 111,1,Answer exten = 111,n,MeetMe(111,pdMX) exten = 111,n,Hangup I want to redirect both channels to the conference 111. What's wrong it? With Regards, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tampa Asterisk User Group Meeting Monday
Hello, We will be having another Tampa Bay area Asterisk User Group meeting on Monday, February 5th at 7PM All Asterisk users from newbies to Gurus are encouraged to attend. For more information visit our website: http://asteriskpbx.meetup.com/1/calendar/5394922/ Thanks, MATT--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1'
As everybody must be watching the superbowl. I post this to let you have some fun while thinking what this can be. TDM400p (fxo) connected via loopstart to ports in an AvayaG3 call comes in from the avaya to the tdm card: WARNING[27249]: chan_zap.c:6299 ss_thread: CallerID returned with error on channel 'Zap/4-1' but call can be processed normally. comments? -- Erick Perez ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Java FastAGI implementation has the most market share?
Steve, keep me in touch please ? We are also looking for moving all our activities to java platform. Let me know if You'll find/test something like asterisk2billing written in java ? Cheers, Kate On 2/1/07, Steve Prior [EMAIL PROTECTED] wrote: When I was looking for a Java FastAGI interface for Asterisk I came across asterisk-java first and didn't realize there was more than one out there. It seems to work fine and I've got my first project working with it, but I was wondering which Java FastAGI implementation is the most popular and how they compare against each other. So I'm aware of: asterisk-java JastAGI OrderlyCalls Any comments on who the front runner is and why? Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help - Received response: Forbidden from 'Unknown
I have a weird problem
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext - Panasonic Ext No Problems
Panasonic Ext - SIP Ext No Problems
SIP Ext - VOIP Provider No Problems
Panasonic Ext - VOIP Provider Errors
-- Working SIP - VOIP
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/610-097aee60,
SIP/acevoip/03) in new stack
-- Called acevoip/03
-- SIP/acevoip-097b52c0 is making progress passing it to
SIP/610-097aee60
-- SIP/acevoip-097b52c0 is making progress passing it to
SIP/610-097aee60
== Spawn extension (from-sip, 903, 1) exited non-zero on
'SIP/610-097aee60'
-- Not Working Panasonic Ext - VOIP
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/31-1,
SIP/acevoip/03) in new stack
-- Called acevoip/03
[Jan 29 11:00:36] WARNING[20642]: chan_sip.c:11731
handle_response_invite: Received response: Forbidden from 'Unknown
sip:[EMAIL PROTECTED];tag=as3a292a14'
-- SIP/acevoip-097b1358 is circuit-busy
--
Both numbers dialled were exactly the same (9 is the leading number on
all calls in the system and is stripped before dialing), I just replaced
the numbers with .
Tested from several different sip phones and Panasonic handsets, and it
is only with outgoing calls to VOIP, incoming that go to a Pana
extensions work fine.
--- Extensions.conf
[dialstring]
exten = t,1,Dial(Zap/g1/100,60,tn)
exten = i,1,Dial(Zap/g1/100,60,tn)
[from-e100p]
include = dial-sip
include = out-voip
[dial-e100p]
exten = _1XX,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_1XX,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(
num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN})
exten = _1XX,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _1XX,4,Dial(Zap/g1/${EXTEN},90,r)
exten = _91XX,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_91XX,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID
(num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = _91XX,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _91XX,4,Dial(Zap/g1/${EXTEN:1},90,r)
exten = _9X.,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_9X.,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(
num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = _9X.,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _9X.,4,Dial(Zap/g1/${EXTEN},90,r)
exten = _9X.,5,Busy
exten = 000,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
000,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(n
um)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN})
exten = 000,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = 000,4,Dial(Zap/g1/000,60,r)
exten = 9000,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
9000,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(
num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = 9000,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = 9000,4,Dial(Zap/g1/000,60,r)
[out-voip]
exten = _902X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _903X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _905X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _906X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _908X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _954X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _955X.,1,Dial(SIP/acevoip/${EXTEN:1})
[from-acevoip]
include = dialstring
exten = 073...,1,Answer
exten = 073...,2,Dial(Zap/g1/100,60,tn)
exten = _073.XX,1,Answer
exten = _073.XX,2,System(mkdir
/mnt/data/Recording/${SIP_HEADER(TO):12:3})
exten =
_073.XX,3,Set(CALLFILENAME=/mnt/data/Recording/${SIP_HEADER(TO):12:3
}/${SIP_HEADER(TO):12:3}-Received-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-$
{CALLERID(num)})
exten = _073.XX,4,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _073.XX,5,Dial(SIP/${SIP_HEADER(TO):12:3},60,tn)
exten = _073.XX,6,Voicemail(${SIP_HEADER(TO):12:3}u)
exten = _073.XX,7,Hangup
exten = _073.XX,106,Voicemail(${SIP_HEADER(TO):12:3}u)
exten = _073.XX,107,Hangup
include = dial-sip
include = dial-e100p
[from-sip]
include = dialstring
include = dial-sip
include = out-voip
include = dial-e100p
[dial-sip]
exten = 600,1,Dial(Zap/g1/100,60,tr)
exten = 9600,1,Dial(Zap/g1/100,60,tr)
exten = _6XX,1,SetMusicOnHold(random)
exten = _6XX,2,System(mkdir /mnt/data/Recording/${EXTEN})
exten =
_6XX,3,Set(CALLFILENAME=/mnt/data/Recording/${EXTEN}/${EXTEN}-Received-$
{STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${CALLERID(num)}.wav49)
exten = _6XX,4,MixMonitor(${CALLFILENAME}|v(0)V(0))
exten = _6XX,5,Dial(SIP/${EXTEN},45,Ttr)
exten = _6XX,6,Voicemail(u${EXTEN})
exten = _6XX,7,Hangup
exten = _6XX,106,Voicemail(b${EXTEN})
exten = _6XX,107,Hangup
exten = _96XX,1,SetMusicOnHold(random)
exten =
Re: [asterisk-users] TDM400 noHangup
No, that is just a tone. Correct disconnect supervision is an electrical thing. Either reversing the polarity or dropping battery. Mochamad Susantok wrote: I am don't sure what PBX we have, but when i hangup i hear like tut tut tut in the other side, Is that not disconnected signal ? FYI when i take call from PBX to VoIP client or vice versa. it's ok. What do you think, are there is have some relation with my problem ? What PBX do you have connected to the Asterisk servers? My experience is that most PBXs do not provide a disconnect signal on their analog station ports. I have had the most success with disconnects on Avaya PBXs. Nortel analog stations last I tested with any, did not provide a disconnect signal. On these other PBXs you will need to use band-aids like having zaptel monitor for busy or dial tone when the analog station hangs up. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap FXS slow to reset?
From: Scott Walde [EMAIL PROTECTED] Yuan LIU wrote: exten = s,n,Dial(Zap/1SIP/202SIP/203,18) exten = s,n,Dial(Zap/1SIP/201SIP/202SIP/203,42) Interestingly, although the Asterisk Manual (by Mark Spencer and so on) contains an almost identical sample plan fragment, in reality, it seems to need a Wait() in between to reset the Zap channel in the first Dial(). I tested with TDM400. Inserted a 1-2 sec wait and it did what you wanted. I thought about doing that but was worried about the condition where someone answers a phone during that 1 second (which is actually quite likely to happen) and they only get dialtone rather than answering the incoming call. Has this not been an issue for you? ttyl srw I'm not using this in production. But this should count as a bug IMO, either in Zaptel or in the card. (Especially because the manual cites such use.) I tested a workaround: add a NoOp() in between; Zaptel still gives an error, but the Zap channel rings afterward. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which Java FastAGI implementation has the most market share?
Kate Kretz wrote: Steve, keep me in touch please ? We are also looking for moving all our activities to java platform. Let me know if You'll find/test something like asterisk2billing written in java ? I haven't received any feedback at all on the relative use of the java options, but I'm pretty happy with the way a little project turned out in asterisk-java. My project was to see how well asterisk-java would work in combination with Lumenvox to create a speech enabled AGI, so just for kicks I've ported their Pizza ordering demo to Java using it. In the process I've been working with Lumenvox to fix the couple of problems which turned up as a result of this experiment, and use an asterisk-java code change which is available in their latest svn. Sometime soon my code will be made available most likely through the Lumenvox site so others can use it as a starting point. Overall I'll say that I really like using Java to control such a dial plan. In this particular case the output is a simple pizza order which I've modeled as a plain old Java object (POJO), so once the dial plan has built up the object it can simply be passed to whatever back end (possibly J2EE) code which processes the transaction without regard for the user interface that created it. Sounds very maintainable to me. I did the development/test right in the Eclipse IDE and could use the debugger when necessary - I've got to believe that's better than trying to trace through a regular dial plan. I also really like the fact that aside from sound files and just a couple of lines of dial plan code to call the Java, all the actual Java code is running in a different server box so I'm keeping the load down on my Asterisk box and have flexibility in where I deploy things. Steve Cheers, Kate ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Continue line in config files?
From: C F [EMAIL PROTECTED] how would a line be soo loogg? It doesn't take a very complicated expression to go over 80 characters. Also consider multiple voice files in PlayBack() or Background(), System() calls, etc. Yuan Liu On 2/4/07, Yuan LIU [EMAIL PROTECTED] wrote: Is there anything that allows a logical line to extend to the next physical line? Printed files are so hard to read with blind line wraps - and my printer doesn't even automatically wrap. Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Does TE212P card work on HP DL380 G5?
Hi all, I am preparing the new asterisk system for 60 concurrent calls with 2 E1. I have to use server HP DL380 G5. Anybody get TE212P card work on this server using asterisk? Thanks, M ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help - Received response: Forbidden from 'Unknown
Very odd.
My guess is that it's one of 2 things.
Slightly different number being sent to the SIP provider. (unlikely)
Different callerid being sent to the SIP provider.
Have you tried blanking the callerid before making the outbound call?
(in case the provider doesn't like it)
PaulH
On Mon, 2007-02-05 at 14:08 +1000, James's Asterisk wrote:
I have a weird problem….
Asterisk 1.4
E100P connected to a Panasonic TDA phone system
Here is what I get
SIP Ext - Panasonic Ext No Problems
Panasonic Ext - SIP Ext No Problems
SIP Ext - VOIP Provider No Problems
Panasonic Ext - VOIP Provider Errors
-- Working SIP - VOIP
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/610-097aee60,
SIP/acevoip/03……..) in new stack
-- Called acevoip/03……..
-- SIP/acevoip-097b52c0 is making progress passing it to
SIP/610-097aee60
-- SIP/acevoip-097b52c0 is making progress passing it to
SIP/610-097aee60
== Spawn extension (from-sip, 903…….., 1) exited non-zero on
'SIP/610-097aee60'
-- Not Working Panasonic Ext - VOIP
-- Executing [EMAIL PROTECTED]:1] Dial(Zap/31-1,
SIP/acevoip/03……..) in new stack
-- Called acevoip/03……..
[Jan 29 11:00:36] WARNING[20642]: chan_sip.c:11731
handle_response_invite: Received response: Forbidden from 'Unknown
sip:[EMAIL PROTECTED];tag=as3a292a14'
-- SIP/acevoip-097b1358 is circuit-busy
--
Both numbers dialled were exactly the same (9 is the leading number on
all calls in the system and is stripped before dialing), I just
replaced the numbers with ……...
Tested from several different sip phones and Panasonic handsets, and
it is only with outgoing calls to VOIP, incoming that go to a Pana
extensions work fine.
--- Extensions.conf
[dialstring]
exten = t,1,Dial(Zap/g1/100,60,tn)
exten = i,1,Dial(Zap/g1/100,60,tn)
[from-e100p]
include = dial-sip
include = out-voip
[dial-e100p]
exten = _1XX,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_1XX,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN})
exten = _1XX,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _1XX,4,Dial(Zap/g1/${EXTEN},90,r)
exten = _91XX,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_91XX,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = _91XX,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _91XX,4,Dial(Zap/g1/${EXTEN:1},90,r)
exten = _9X.,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
_9X.,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = _9X.,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _9X.,4,Dial(Zap/g1/${EXTEN},90,r)
exten = _9X.,5,Busy
exten = 000,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
000,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN})
exten = 000,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = 000,4,Dial(Zap/g1/000,60,r)
exten = 9000,1,System(mkdir /mnt/data/Recording/${CALLERID(num)})
exten =
9000,2,Set(CALLFILENAME=/mnt/data/Recording/${CALLERID(num)}/${CALLERID(num)}-Called-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${EXTEN:1})
exten = 9000,3,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = 9000,4,Dial(Zap/g1/000,60,r)
[out-voip]
exten = _902X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _903X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _905X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _906X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _908X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _954X.,1,Dial(SIP/acevoip/${EXTEN:1})
exten = _955X.,1,Dial(SIP/acevoip/${EXTEN:1})
[from-acevoip]
include = dialstring
exten = 073…….,1,Answer
exten = 073…….,2,Dial(Zap/g1/100,60,tn)
exten = _073…..XX,1,Answer
exten =
_073…..XX,2,System(mkdir /mnt/data/Recording/${SIP_HEADER(TO):12:3})
exten =
_073…..XX,3,Set(CALLFILENAME=/mnt/data/Recording/${SIP_HEADER(TO):12:3}/${SIP_HEADER(TO):12:3}-Received-${STRFTIME(${EPOCH},,%d%m%Y-%H%M%S)}-${CALLERID(num)})
exten = _073…..XX,4,MixMonitor(${CALLFILENAME}.wav49|v(0)V(0))
exten = _073…..XX,5,Dial(SIP/${SIP_HEADER(TO):12:3},60,tn)
exten = _073…..XX,6,Voicemail(${SIP_HEADER(TO):12:3}u)
exten = _073…..XX,7,Hangup
exten = _073…..XX,106,Voicemail(${SIP_HEADER(TO):12:3}u)
exten = _073…..XX,107,Hangup
include = dial-sip
include = dial-e100p
[from-sip]
include = dialstring
include = dial-sip
include = out-voip
include = dial-e100p
[dial-sip]
exten = 600,1,Dial(Zap/g1/100,60,tr)
exten =
[asterisk-users] Local hangup after Dial()?
Another dumb question: Can a dial plan continue after local hangup when using Dial()? For example, [incoming] exten = s,1,Dial(Zap/1) exten = s,2,Congestion() exten = s,3,Hangup() --- Asterisk seems to insist that a dial plan is complete when Zap/1 hangs up and do not go into priorities 2 and 3. Any idea? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Local hangup after Dial()?
Yuan LIU wrote: Another dumb question: Can a dial plan continue after local hangup when using Dial()? For example, [incoming] exten = s,1,Dial(Zap/1) exten = s,2,Congestion() exten = s,3,Hangup() --- Asterisk seems to insist that a dial plan is complete when Zap/1 hangs up and do not go into priorities 2 and 3. Use the h extension. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
