Re: [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones
Hello, Just FYI, we have a small asterisk with 4 x100p cards in a cheap motherboard and Athlon processor for our office. There is no way to avoid irq sharing, and we have the ocassional 'chirp' noise. This is the /proc/interrupts output: # cat /proc/interrupts CPU0 0:1449479 XT-PIC timer 1: 1211 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 12428957 XT-PIC wcfxo 5: 12429375 XT-PIC wcfxo 8: 5 XT-PIC rtc 10: 13250633 XT-PIC eth0, wcfxo 11: 12428834 XT-PIC wcfxo 12:278 XT-PIC PS/2 Mouse 14: 21213 XT-PIC ide0 NMI: 0 LOC:1449417 ERR: 5283 MIS: 0 Now I have just setup an old Powermac 9600, its replacing the athlon box. It is up and running just for 10 minutes. It works fine, and the /proc/interrupts looks much better: cat /proc/interrupts CPU0 2: 0 PMAC-PIC Edge MACE-txdma 3: 67851 PMAC-PIC Edge MACE-rxdma 12: 7 PMAC-PIC Edge 53C94 13: 51368 PMAC-PIC Edge MESH 14: 63266 PMAC-PIC Edge MACE 15: 0 PMAC-PIC Edge SCC 16: 0 PMAC-PIC Edge SCC 18: 2769 PMAC-PIC Edge ADB 19: 0 PMAC-PIC Edge SWIM3 24:2164186 PMAC-PIC Level wcfxo 25:2164002 PMAC-PIC Level wcfxo 27:2163854 PMAC-PIC Level wcfxo 28:2163665 PMAC-PIC Level wcfxo BAD: 0 For a small PBX a used powermac might work fine... And I still have one pci slot to spare... On Wed, 2004-08-25 at 13:57, spectro wrote: IMHO, If you plan to use analog phones the cheapest is to buy a bunch of sipuras instead of TDM40B. (TDM40B = 4 FXS for $300, $75 each; sipura SPA2000 = 2 FXS for $100, $50 each) Order one TDM04B (4 FXO) and 1 Sipura 2000 for each 2 analog extensions. You can also mix it up, lets say drop two incoming lines and order a 2 FXS, 2 FXO TDM instead. Then subscribe to a VoIP provider like Voicepulse Connect to dial-out through IAX. Use the FXS on the TDM for fax machines and make these the only to dial-out through the analog outgoing lines. Of course, I would rather buy some IP phones instead of analog ones through FXS. For your remote office, depending on the number of extensions, you can either setup a small asterisk box or just use Sipuras 3000 and 2000 connected to your main office's asterisk server. On Tue, 24 Aug 2004 17:11:36 -0600, Andrew Elchuk [EMAIL PROTECTED] wrote: Hi I am interested in setting up an Asterisk PBX in my office with digium hardware, and I just have a few questions in regards to what I would need. It is my understanding that an FXO card is used to interface with an incoming/outgoing phone line, and an FXS card is used for interfacing with a phone within the system. Currently we have 4 incoming/outgoing phone lines and would like to have 20 phones in the system. In order to accomodate this, would you either reccommend having 1 TDM04B (4 FXO modules on it) for the 4 incoming/outgoing lines, and 5 TDM40B (4 FXS modules on each) for the 20 phones we would have in the system. Or would you reccommend 1 TDM04B for the 4 incoming/outgoing lines, and a T100P connected to a channel bank of some sort to connect to the internal phones? If you reccommend the T100P and channel bank, where do you suggest I get an FXS channel bank? Please let me know if I got any of this mixed up (like if I got the FXO and FXS cards mixed up) and thank you in advance for your help in us deciding the hardware we need for a new PBX. Andrew Elchuk P.S. We are currently using an X-Like software phone with a free world dialup account for communication with our other office in a different city. My question is if I can configure the extensions.conf to connect to a free world dialup number when executing a dial command, or would I need to edit sip.conf as well or some other configs or is that even possible? Thank you again. -- Andrew Elchuk Technical Associate Cronus Technologies 248 - 111 Research Drive Saskatoon, SK S7N 2X8 Tel: (306) 652-5798 ext. 112 Fax: (306) 652-5799 Toll Free: 1-877-655-5798 http://www.cronustech.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L
Re: [Asterisk-Users] Distinctive Ring Cadences
On Wed, 2004-08-25 at 20:38, Chris Shaw wrote: Cool! I could see this being very useful, for example you could have an IVR that says something like Please set the priority of your call, 1 for urgent, 2 for normal or 3 for low then if 1, bellcore-r4, if 2 bellcore-r3, if 1 bellcore-r1! What for? People will allways hit 1 g -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error compiling meetme2
Hi Geoff, Geoff Nordli wrote: I was able to compile the module and it loads correctly, but I am still having problems with the app. I see all the users in the conference, but I can't kick them out, or change their mode from talk to listen-and-talk. No errors are showing up anywhere. I am not really sure how to troubleshoot this, any ideas? Thanks, Enable register_globals in php. You can also put an extract($_GET); in the top of the php file. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multiple SIP phones ringing for same extension
Hi, David Gurr wrote: Can someone confirm what I should expect the correct behaviour to be on incoming calls if I have multiple SIP phones configured for the same username? I'd expect all the phones registered under the username that that extension is associated with to ring, and the first one that answers gets it. What I get, is just the first phone that registered gets a ring. The second one doesn't ring at all. Asterisk won't work this way. Just the last phone registered will ring. There was a big thread a month ago in this list and a Bounty placed for adding the feature. Search for sip simultaneous in google or the wiki (http://www.voip-info.org) -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call stealing
Ben Merrills wrote: Hi, How can I (through the manager interface) steal a call from one phone, and transfer it to another? Does asterisk provide for actions like this? Its a common action in Lucent systems it seems. Cheers, Ben You can use the Redirect command. Visit http://www.asternic.org and look at the Flash Operator Panel. It can do that and more.. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Hi Francis, Francis Augusto Medeiros wrote: Hi there everyone! I work at an office where we plant to have about 12-15 phone extensions. Costs of PBX are cheaper, but they are not expandable and, as the office is brand new, I want to use all modern stuff. My question is: if I buy 12-15 Grandstream Budgetone 101 phones, and install and asterisk server, as well as a Digium TDM400 for POTS access, will I have the same voice quality and standards as a PBX-only, with traditional phones? Or should I go all the way to Digium's TDM? Or should I forget the whole thing and get a traditional PBX? ;) If you already have the analog telephone wiring in place, and you are on a budget, I recomend you to use sipura spa-2000 adapters. They are a whole lot better than GS phones. You can have 3way conferences and attendant transfers. With GS you cannot do that. The price is as good for the sipuras as the GS phones, about $50 per FXS port, plus a cheap analog phone and you will be all set. My concerns are most latencies. Our network will be a switch with lots of ports, all 100mb/s, with VERY low traffic. Internal calls (SIP to SIP) will sound great. You will probably experience some echo when going to POTS. I did not try the Sipura SPA-3000 yet, but it seems to be a cheap alternative to a gateway, providing you with one FXO and one FXS for $130 or so. the echo cancellation in the sipura works well for fxs, it might work well to for fxo. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help - is voip good for in-house calls?
Andrew Kohlsmith wrote: On Sunday 15 August 2004 12:03, Nicolas Gudino wrote: If you already have the analog telephone wiring in place, and you are on a budget, I recomend you to use sipura spa-2000 adapters. They are a whole lot better than GS phones. You can have 3way conferences and attendant transfers. With GS you cannot do that. The price is as good for the sipuras as the GS phones, about $50 per FXS port, plus a cheap analog phone and you will be all set. Why on earth would you install SPA-2000s and endure that wiring mess? An FXS channel bank and a BIX strip will save you YEARS in lost time due to wiring and general messiness! I prefer the wiring mess and sipuras than the GS phones. That's all. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF after answer
Hello, Marc C Storck wrote: I always have an browser window with the wiki open, but i couldn't find what i need: 1) Someone calls 123456 on my PRI 2) asterisk sees the call 3) asterisk dials a number 4) the number called by asterisk get answered 5) asterisk waits 10 secs and sends DTMF 6) asterisk connects both calls Marc Excerpt from show application dial: 'D([digits])' -- Send DTMF digit string *after* called party has answered but before the bridge. (w=500ms sec pause) So: Exten = 1,1,DIAL(SIP/[EMAIL PROTECTED],90,D(w9876543) Will dial the number, wait a few seconds, and send DTMF 9876543 before bridging the call with the calling party. Best regards, -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E1 monochannel :-(
Hola Horacio,
Comentarios en línea...
Horacio J. Peña wrote:
Hola!
I'm using asterisk as H.323 - PRI gateway. First call goes
thru ok, second concurrent call fails with:
Aug 6 11:52:30 DEBUG[737292]: chan_h323.c:1038 setup_incoming_call: Sending to
context [ip2pri]
-- Executing Dial(H323/ip$192.168.32.25:60271/984, Zap/1/9541163107100) in new
stack
Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to create channel of
type 'Zap'
Aug 6 11:52:30 NOTICE[753677]: app_dial.c:554 dial_exec: Unable to create channel of
type 'Zap'
== Everyone is busy at this time
Aug 6 11:52:40 WARNING[753677]: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in
context 'ip2pri'
Aug 6 11:52:40 WARNING[753677]: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't' in
context 'ip2pri'
Aug 6 11:52:40 DEBUG[81926]: chan_h323.c:1179 cleanup_connection: Cleaning up our mess
My configs are:
h323.conf:
[general]
port = 1720
bindaddr = 0.0.0.0
disallow=all
allow=g729
gatekeeper = DISABLE
context=ip2pri
[ip2pri] ; is this needed?
type=user
context=ip2pri
extensions.conf:
[general]
static=yes
writeprotect=yes
[globals]
[ip2pri]
exten = _9.,1,Dial(Zap/1/${EXTEN:0}) ; i must send the 9 to the PRI...
^^^
Replace Zap/1 with Zap/g1
Saludos!
--
Nicolás Gudiño
House Internet S.R.L.
Buenos Aires - Argentina
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Re: [Asterisk-Users] Barge in on to agents conversation
Hello, On Wed, 2004-08-04 at 11:35, Navnit Chachan wrote: Hi, 1. When an agent is active on a call, i need the ablity for a third person to join the conversation. Basically barge in by a supervisor, participate in the conversation and then leave. Asternic, the Flash Operator Panel can do this, but you need to open it on a web browser and use your mouse to drag the manager extension to any leg of an already bridged call, with some extensions logic and meetme in the mix. I'm not sure if it will fit your needs, but it might help... http://www.asternic.org Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco SIP Phone 7960 DTMF Problem
Hello, On Wed, 2004-08-04 at 04:35, Miroslav Nachev wrote: Hi, When we use BudgeTone where the DTMF is set to via RTP (RFC2833) all the DTMF functionality of Asterisk is working OK. When use Cisco 7960 the transfer is working OK, but when I try to remote pick-up the call through '*8#' I can't do that because the Cisco Phone start busy signal. How can I start using all DTMF features using Cisco Phone? Did you try by dialing just '*8' ? -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Identifying which call an event belongs to
Hello, On Wed, 2004-08-04 at 18:56, Michael Ulitskiy wrote: Hi, I guess I need some help with management interface. I would like to watch calls through the management interface, but I don't know how to identify which call an event belongs to or in other words how to associate a call and uniqueid field of event. Let's say I send the following manager command: action: originate channel: sip/[EMAIL PROTECTED] callerid: 1212555 MaxRetries: 1 WaitTime: 10 Application: AGI Data: callback.agi|212125551212555 Try inserting in your originate command: ActionID: SOME_RANDOM_ID Then I'm receiving the following events: Uniqueid: 1091642334.98 Event: Newchannel Callerid: State: Down Channel: SIP/pbx1-fc4f And you would probably receive: Uniqueid: 1091642334.98 Event: Newchannel Callerid: State: Down Channel: SIP/pbx1-fc4f ActionID: SOME_RANDOM_ID I did not try this, but I know that ActionID is implemented in some manager commands. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Parking SIP Phones
Hi Trevor, Trevor Peirce wrote: Hello, I know not too long ago I saw /something/ _somewhere_ about an adjustment to call parking that allowed blind transfers from SIP phones to park a call and still be able to hear the parking lot stall number. Unfortunately, I have no idea where I saw that (google turned up little, couldn't find it on the list either). I'm using Sipura SPA-2000 adapters and it doesn't seem to work with today's CVS. I can park from sip phones using asterisk transfer key and hear the parking lot with no problems. I'm using Grandstream phones, sipuras, and a recent asterisk from CVS. It does not work if I transfer with the ATA or phone transfer feature. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Hi Eric, On Sat, 2004-07-31 at 17:55, Eric Bart wrote: I don't understand why sipura can do consultative transfer and why grandstream can't. They're both SIP, aren't they ? They use different sip stacks... and yes, they are both sip. Maybe the sipura transfer is using a sip reinvite or some other SIP command. Does the consultative transfer works when the other parties are not attached to a sipura phone (ie when a sipura phone try to make a consultative transfer from a grandstream to a snom) ? It works with ZAP FXO, Sipuras and Grandstream phones. The sipura is able of 3way conferences by itself. The consultative transfer is a kind of 3way conference for the Sipura.. From what you said, I believe that asterisk is not managing these consulative transfers and is not aware of. These are inter-phone communications (peer to peer). Each peer has to understand each other, which is not easy when mixing multiple technologies. I think that the peers only need to understand how to handle a sip invite. Any sip user agent will do. All the magic is done inside the sipura. PS: I'm not affiliated in any way to Sipura. I just like their products. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Hi Eric, On Mon, 2004-08-02 at 16:26, Eric Bart wrote: It works with ZAP FXO, Sipuras and Grandstream phones. The sipura is able of 3way conferences by itself. The consultative transfer is a kind of 3way conference for the Sipura.. So it seems that the others parties keep running through the sipura, even in a consultative transfer. So you can't have too many transfers going on, it's not suitable for an operator. Is it ? Sipura is limited to 3way conferences (or 2 line appearences) If you would like to have many calls onhold/waiting, you can use asterisk with parking or valet, or even call queues. If you can afford the hardware, you can try with a high end cisco phone. Regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Hello, Eric Bart wrote: Flash don't work for sip This affirmation is too broad, it might not work with X-lite, but flash will work with may sip devices, including cheap ones (grandstreams, sipuras, etc). From: Jozeph Brasil [EMAIL PROTECTED] I have one X100P installed with two SIP extensions using X-Lite, I just would like to transfer the call to another SIP extension; Just a Flash+Extension+Hangup CALL... -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Hi Eric, Eric Bart wrote: Thanks for the correction I didn't know that SIP would do. As I understood the R key will send the flash signal. However does it really act as a transfer ? For the zap transfer, as said in : http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20zap%20transfer when the transferer hangs up each parties are disconnected. This is for ZAP channels, the original question was: have one X100P installed with two SIP extensions using X-Lite, I just would like to transfer the call to another SIP extension; Just a Flash+Extension+Hangup CALL He wants to transfer a call from one SIP extension to another... All sip devices that I know off (I'm not talking about soft phones, I do not use them, so I can say anything about them) have a way to transfer a call to another sip device by themselves (without the help of asterisk). Grandstream phones have a 'transfer' key. If you press that key and then dial the extension you like to transfer and then hangup (just like the original poster asked), it will just work. Its a blind transfer, and you better dial the desired extension right, because if you made a mistake, the call will be lost in limbo as some other users are reporting (a grandstream feature/bug) Sipuras can do this to: just by flashing the analog phone. They are capable of consultative transfers also (they let you talk to the destination party before transferring the call) I tried them both, transferring an inbound call from a ZAP FXO line to a sip extension and it works great, no hangups, no problems. With sipuras I can do consultative transfers also, I use them all the time. You can also achieve the same results by using asterisk transfer feature (T or t options in the dial command). In this case the transfer will be allways blind. It works perfect with ZAP FXO and SIP FXS for me. If you want consultative transfers with asterisk, you can sort of have it by using parking: you can dial '#' to transfer, then send the call to the parked calls extension, and the parked extension will be read back to you. Then you hangup and talk to the extension you want the call to be transferred: 'you have Bob on the extension 702'. The other party can now dial that extension and talk to Bob. Its not a consultative transfer as regular phone users are accustomed, but it works. And if the parked call times out, it will ring back the extension that parked it on the first place. And I'm sure it works also with other technologies as IAX2 or CAPI. Is it what you are experiencing ? With my app when the transferer hangs up the others parties stay connected ... I'm wondering whether it's useful or not :) Maybe your application cann fill the gap for sip devices that are not capable of consultative transfers by themselves... Best regards, -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to IP-PBX
Hello, On Fri, 2004-07-30 at 15:39, Duraid Abbas wrote: I have a D/41JCT-LS Dialogic board and I want to use it as an IP-PBX. I'm new to IP Telephony and telephony and general and I researched a lot but still confused about what I really need. I know that I can setup an IP-Telephony for my LAN using a SIP server and SIP compatible software phones. But the challenge is how can I connect to the PSTN so that I can send and receive calls? Asterisk will do a wonderfull job as a soft PBX, but my advice is to use hardware from Digium to connet to the PSTN (FXO or T1/E1) and to connect regular analog phones (FXS or T1/E1+ChannelBank): http://www.digium.com/index.php?menu=hardware_products Before purchasing hardware, you can try to set up Asterisk just with SIP softphones and get it to know the platform. Once you are comfortable you can jump on buying some hardware. If you do not have time to investigate yourself search for Asterisk consultants on http://www.voip-info.org Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk GUIs at Astricon * REMINDER *
Holger Schurig wrote: What I'm thinking of is giving each GUI a slot of 10-15 minutes for a presentation and then a panel discussion on the GUI theme. No chance for me to pay flight + entry to conference. My wife would hack me in little pieces :-) Me neither... -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk - stanaphone?
Hi, Zdenek Bouresh wrote: Jerry Glomph Black wrote: snip Perhaps they are fingerprinting and blocking Asterisk access (I hope not). They do not answer their support mail, or questions on their own forum. snip Their service has downtimes very often. But if they really decide to block asterisk , just goto /channels/chan_sip.c and change #define DEFAULT_USERAGENT Asterisk PBX to whatever user agent you want , even their own . Thats it. You don't need to modify the source to change the useragent. Just put: useragent=cisco_super_phone in the general section of sip.conf -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Large Enterprises using asterisk
Hello, avizion wrote: This is exactly what I will be looking for in near future. Our current setup (Old Ericsson PBX) with these system phones having hotkeys for transfer, hold, ACD in/out, multiple lines, etc. and a quite handy feature... the LED that tells my weather a certain agent is busy or not. Soft Phones like IaxClient have some of this - but far from complete imho. I have been discussing this with some managers and they are open for helping to commit hours in terms of development in the open source community. I would be very interested in pointers to development in this area - and even discussion lists for starters. I know the WiKi is a great start but I hope it's missing some linke :) PS: If already existing soft (and/or hard) phones have more of this functionality - please let me know. There are utilities to show that information (led that tells when an agent is busy or not), but not soft phones as fas as I know. Some companies are developing SIP addons to their phones also. Search for asterisk gui on the wiki. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP register and unregister events via Manager API
Hi John, John Todd wrote: At 10:58 AM +0200 on 7/23/04, Holger Schurig wrote: Okay, I have finished my patch. With qualify=yes in sip.conf it looks like this: output snip Event: PeerStatus Peer: SIP/weckhardt PeerStatus: Reachable Time: 81 some more snip - Without the quality, you still get the PeerStatus: Registered and PeerStatus: Unegistered events. John, you can do your color-coding :-) [snip] Not me! :-) I'd point a finger at Nicolás Gudiño and have him include it in the Asterisk Flash Operator panel, which seems to be one of the appropriate places that this could create a graphical representation of registration status and quality= response time. Maybe a red-to-green spectrum of colors on the button background. I'd expecet that each button would need to have probably independent configurations, since some devices may be very far away and thus have different numeric values mapped to different colors. If the device falls out of registration, then perhaps have thin black lines diagonally through the button, and dim it slightly? Thats me... :) Well, we already have in the panel dimmed buttons for SIP peers that are unreachable, and really dimmed ones for not registered ones. Now I will have code the color shift based on the round trip time. Maybe I can zoom out the button instead of color coding? If the latency is high display the button far far away :) -- Nicolás Gudiño House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP register and unregister events via Manager API
Hi Matthias, On Fri, 2004-07-16 at 18:28, Matthias Endler wrote: Hi all, is it possible to receive SIP/IAX register and unregister events via the manager API (like in CLI)? I do receive all kinds of call events (Hangup|Join|Leave|Link|Newchannel|Newexten|Newstate|Rename|Unlink). chan_sip2 supports manager notifications: http://bugs.digium.com/bug_view_page.php?bug_id=759 Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call Intrude
Hi Robb, Robert Boardman wrote: Hi I have looked through the wiki and search the mailing list, but I cannot find a way to intrude on a call, can asterisk do this feature? if so how? If you want to just listen to a call involving a zap channel, you can use ZapBarge. [Synopsis]: Barge in (monitor) Zap channel [Description]: ZapBarge([channel]): Barges in on a specified zap channel or prompts if one is not specified. Returns -1 when caller user hangs up and is independent of the state of the channel being monitored. If you want to barge in on a call and talk to the two parties involved, its possible using a combination of meetme and the manager interface... The Flash Operator Panel I made supports barge in using dragdrop in combination with the meetme E parameter. http://www.asternic.org Best regards, -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New CVS for patch...
Hi Jay, Jay Milk wrote: I want to patch voicemail.c to allow for configurable pager-messages. Looked at the code, and I know I can do that in 10 minutes. Once done, I'm planning to make this patch available to the community, provided the paperwork (release form etc) takes less time than the actual patch. Of course I know that I should based my modification on the latest-available code, but I'm a bit reluctant to upgrade my WORKING asterisk to the latest CVS. Can I rename my asterisk-dir in /usr/src to something different, then check out the latest CVS, make my changes, and if it doesn't work, revert to my working version? Or will Make and its friends throw me for a loop? Yes you can. I do it from time to time. Be sure to remove the contents of /usr/lib/asterisk/modules before installing any version (your current or the latest one), because new modules (if there are any) will not be removed when reverting back to the previous version and you will have problems. And just issue a 'make install' (not a 'make samples'!) -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Delay when dialing with Sipura 2000
Senad Jordanovic wrote: Brian Weaver wrote: I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this? You need to play with: Interdigit Long Timer and Interdigit Short Timer values under Regional tab. Also, make sure you are in advanced mode. Or better yet, look at the dialplan setting in Line1/Line2 (advanced), read the sipura user guide available on the net and learn how to adjust it to match a valid number and send it inmediatly. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Monitoring Asterisk
Glynn Condez wrote: Hi all, I would like to ask if Asterisk will allow to be monitor via web browser. I am planning to create a web interface to monitor the current sip connected end points and status of iax channels use. If i write a code in php to execute this command should it be possible? asterisk -rx iax show channels regards. Its alrady done. And in real time, no need to refresh or post/get to a web page. Its called Flash Operator Panel http://www.asternic.org You can see at a glance: * What extensions are busy, ringing or available * Who is talking and to whom (clid, context, priority) * SIP registration status and reachability * Meetme room status (number of participants) * Queue status (number of users waiting) * Message Waiting Indicator and count * Parked channels You can perform these actions: * Hang-up a channel * Transfer a call leg via dragdrop * Originate calls via dragdrop * Barge in on a call using dragdrop * Set the caller id when transferring or originating a call * Automatically pop up web page with customer details Best regards, -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DISA and AGI: authenticate by caller ID?
Hi Matthew,
Look at the bootom for my recommendation (take note, I did not test it):
On Thu, 2004-07-01 at 14:08, Matthew Simpson wrote:
I want to have a way to authenticate callers to the extension by Caller
ID... if their caller ID is in my database and set to active, they can call
out. [like a calling card but auth'd by CID instead of PIN].
Here is my dialplan:
1234, 1, agi(ldusers.agi)
1234, 2, Hangup
Here is my code:
#!/usr/bin/perl
#
use Asterisk::AGI;
use DBI;
$db = dbname;
$host = hostname;
$port = 3306;
$userid = dbuser;
$password = dpasswd;
$connectionInfo = DBI:mysql:database=$db;$host:$port;
$dbh = DBI-connect($connectionInfo,$userid,$password);
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-answer();
if (my $callerid = $input{'callerid'}) {
$AGI-say_digits($callerid);
$query = SELECT active FROM cids WHERE cid=$callerid;#
active should be 1 if the caller ID is found and set active
$sth = $dbh-prepare($query);
$sth-execute();
$sth-bind_columns(undef, \$active);
$sth-fetch();
if($active)
$AGI-exec('DISA','no-password|disa');
^
Instead of executing the application, try creating a new context in your
dialplan that executes DISA. You can send the call to that context like
this:
$AGI-set_context(disa);
$AGI-set_extension(s);
$AGI-set_priority(1);
}
$AGI-hangup();
exit;
In extension.conf add the disa context like this:
[disa]
exten = s,1,disa,no-password|disa
This way, if an error happens with DISA, it will be displayed at the
asterisk console (it will not be hidden inside AGI).
Good luck,
--
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Re: [Asterisk-Users] voicemail notification?
Hi Rich, On Thu, 2004-07-01 at 11:36, Rich Adamson wrote: Just upgraded to cvs Head this morning and noticed our voicemail notification (via email) is failing with: Jul 1 07:48:38 WARNING[1217669936]: app_voicemail.c:837 sendmail: E-mail addres s missing for mailbox [3000]. E-mail will not be sent. However, a valid address in voicemail.conf has been working just fine until now. Sendmail is running, etc. If I add a second email address (eg, pager), it works but the first address does not, like: 3002 = 3002,Rich,[EMAIL PROTECTED],[EMAIL PROTECTED] Played with the context to ensure that wasn't an issue. Faintly remember seeing something modified via cvs list, but can't seem to find anything addressing this one. Google doesn't provide any hints. Thoughts? Another bug was introduced in function notify_new_message: the event sent to manager does not include the voicemail context, so the manager notifications allways return 0 messages. I will submit a bug/patch to the bugtracker for this (as it affects the MWI in my flash operator panel), and I will try to look also at your problem. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail notification?
Voicemail email notifications are fixed on CVS as of now (thanks to citats). On Thu, 2004-07-01 at 15:16, Nicolas Gudino wrote: Hi Rich, On Thu, 2004-07-01 at 11:36, Rich Adamson wrote: Just upgraded to cvs Head this morning and noticed our voicemail notification (via email) is failing with: Jul 1 07:48:38 WARNING[1217669936]: app_voicemail.c:837 sendmail: E-mail addres s missing for mailbox [3000]. E-mail will not be sent. However, a valid address in voicemail.conf has been working just fine until now. Sendmail is running, etc. If I add a second email address (eg, pager), it works but the first address does not, like: 3002 = 3002,Rich,[EMAIL PROTECTED],[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail notification?
Hi, I have just submited bug 1962. If you are using the Flash Operator Panel (and maybe other swtichboard/manager applications with MWI) with current CVS-HEAD, you might need to apply the patch to get MWI working. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail notification?
On Thu, 2004-07-01 at 18:03, Nicolas Gudino wrote: Hi, I have just submited bug 1962. If you are using the Flash Operator Panel (and maybe other swtichboard/manager applications with MWI) with current CVS-HEAD, you might need to apply the patch to get MWI working. Best regards, Fixed on CVS now.. thanks Mark! -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Flah Operator Panel show iax2 trunk
On Mon, 2004-06-28 at 16:02, Justin Carlson wrote: Thank you for the prompt reply but when I add 7;8;9, in my button number field the iax2 button goes away. i just got .10 today . That feature will be available in 0.11, is not complete yet (I'm working on it). Please subscribe to the operator panel mailing list to continue this thread. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X101P on a UK BT line ---- txgain issue
Hi Richard,
These complex impedances are all supported in the Silabs chips used in
both the new TDM FXO module and the FXS module, but the driver currently
sets them to 600 Ohms.
I guess at some stage a patch will appear to perhaps set these depending
on the default tonezone set in the config files.
This was submited today to CVS (answer to your prays?):
Update of /usr/cvsroot/zaptel
In directory mongoose.digium.com:/tmp/cvs-serv10293
Modified Files:
wcfxs.c
Log Message:
Add support for international impedence matching (improves echo abroad!)
Index: wcfxs.c
===
RCS file: /usr/cvsroot/zaptel/wcfxs.c,v
retrieving revision 1.73
retrieving revision 1.74
diff -u -d -r1.73 -r1.74
--- wcfxs.c 23 Jun 2004 18:24:21 - 1.73
+++ wcfxs.c 25 Jun 2004 14:34:07 - 1.74
@@ -28,7 +28,6 @@
#include linux/errno.h
#include linux/module.h
#include linux/init.h
-#include linux/usb.h
#include linux/errno.h
#include linux/pci.h
@@ -90,6 +89,95 @@
{43,LOOP_CLOSE_TRES_LOW,0x1000},
};
+static struct fxo_mode {
+ char *name;
+ int ohs;
+ int ohs2;
+ int rz;
+ int rt;
+ int ilim;
+ int dcv;
+ int mini;
+ int acim;
+} fxo_modes[] =
+{
+ { FCC, 0, 0, 0, 0, 0, 0x3, 0, 0 },/* US, Canada */
+ { TBR21, 0, 0, 0, 0, 1, 0x3, 0, 0x2 },/* Austria, Belgium,
Denmark, Finland, France, Germany,
+
Greece, Iceland, Ireland, Italy, Luxembourg, Netherlands,
+
Norway, Portugal, Spain, Sweden, Switzerland, and UK */
+ { ARGENTINA, 0, 0, 0, 0, 0, 0x3, 0, 0 },
+ { AUSTRALIA, 1, 0, 0, 0, 0, 0, 0x3, 0x3 },
+ { AUSTRIA, 0, 1, 0, 0, 1, 0x3, 0, 0x3 },
+ { BAHRAIN, 0, 0, 0, 0, 1, 0x3, 0, 0x2 },
+ { BELGIUM, 0, 1, 0, 0, 1, 0x3, 0, 0x2 },
+ { BRAZIL, 0, 0, 0, 0, 0, 0, 0x3, 0 },
+ { BULGARIA, 0, 0, 0, 0, 1, 0x3, 0x0, 0x3 },
+ { CANADA, 0, 0, 0, 0, 0, 0x3, 0, 0 },
super big snip
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Re: [Asterisk-Users] Busy message
Hi Keith Keith Waters wrote: There are other users running the latest CVS-HEAD reporting that problem (asterisk segfaults when unable to create channel). Maybe you have to revert to a previous version till the bug is fixed. ( cvs -D ) OK, thanks, will try that (btw, cvs -D is an invalid command) 'cvs -D' is incomplete, you have to specify the date of the version you are requesting after the 'D'. Anyways, it seems that the problem is fixed on CVS. Do a 'cvs update' -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Busy message
Hi Keith,
Hi All... I'm a newbie, just busy getting to grips with asterisk.
I've set up the following, but it causes a segfault when I call somebody who
is offline:
exten = _[123456789],1,NoOp(.call for .${EXTEN})
exten = _[123456789],2,Dial(SIP/${EXTEN},60,tr)
exten = _[123456789],3,Voicemail(u${EXTEN})
exten = _[123456789],103,Hangup
I get...
-- Executing NoOp(SIP/54321-b373, .call for .12345) in new stack
-- Executing Dial(SIP/54321-b373, SIP/12345|60|tr) in new stack
Jun 22 13:37:58 NOTICE[13326]: app_dial.c:681 dial_exec: Unable to create
channel of type 'SIP'
Segmentation fault
Are you running Redhat or Fedora? If so, read this thread for a solution:
http://lists.digium.com/pipermail/asterisk-users/2004-January/031953.html
--
Nicolas Gudino
House Internet S.R.L.
Buenos Aires - Argentina
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Re: [Asterisk-Users] Busy message
Keith Waters wrote: Are you running Redhat or Fedora? If so, read this thread for a solution: http://lists.digium.com/pipermail/asterisk-users/2004-January/031953.html Nope, SUSE SLES 8 There are other users running the latest CVS-HEAD reporting that problem (asterisk segfaults when unable to create channel). Maybe you have to revert to a previous version till the bug is fixed. ( cvs -D ) -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax detected, but no fax extension
Hi Patrick Patrick J. Conroy wrote: Hello all, I have a fax machine attached to one of the FXS ports on my channel bank running into one of the spans of my TE405P. Every time I try to send a fax, I get the error Fax detected, but no fax extension in asterisk. Does anyone know why this would happen? The only other reference I have found that relates to this in the list said to enable OLD_DSP_ROUTINES and rebuild and reinstall asterisk. I have done that, but there is no change. If you used CVS-HEAD there is a new faxdetect parameter for zapata.conf . I have not tried, but it might solve your problem. ;faxdetect=both ;faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Controlling SIP mobile extensions.
Hi, XISCOAIR wrote: Hi everybody, I'm trying to develop a web application for controlling if SIP users are registered in * or not, and show it in a web. My problem is that I don't now if it's possible to do a Shell Script to control this: 1. Connect to console. 2. Execute command. 3. Obtain users registered. 4. Update a BdD. This is possible? There are any best way to implement this? Thanks a lot. It can be done, in fact it's already done. Look here: http://www.voip-info.org/tiki-index.php?page=Asterisk%20GUI Monastery does exactly what you describe and a bit more. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura-SPA2000 background noise
Hi Brian, Brian Cuthie wrote: BTW, anyone know how to get the SPA-2000 do drop loop current momentarily when the other end hangs up? -brian There is a web configuration option to reverse the polarity in the latest 2.0 firmware. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Receptionist manager program.
Hi, On Tue, 2004-06-01 at 12:24, Jonathan Moore wrote: Very definetely interested in this. I can think of a lot of clients that would like it as well. The astgui guys claim there is a potential problem with the manager interface access. They have written some kind of demon to manage access. Have you seen and issue similar to this or seen a problem with greater than 1000 manager accesses per day? I have not experienced any problems, but my setup is small. The Flash Operator Panel (my baby :) ) has a daemon that interfaces to the manager port, so there is only one connection open for it, but I also have a web page that access the asterisk manager port in a regular basis (for agents login/logout), and I don't have problems or crashes. I'm running CVS-HEAD. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Downgrading Asterisk
Hi Nik Nik Martin wrote: I upgraded to the latest HEAD version of asterisk, and all IAX calls started sounding choppy. It was suggested on the IRC channel that I go back to asterisk -stable to determine if that fixes it. Is downgrading as simple as upgrading? Because now, -stable builds fine, but I get an error on the asterisk console when starting, something about ast_get_txt not found. Recompiling and installing asterisk HEAD afterwards works just fine. snip Any ideas? Try deleting or moving all the modules from /usr/lib/asterisk/modules before performing the make install with the old version. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P problems with 1 FXS, 1 FXO
Hi, On Wed, 2004-05-19 at 14:51, David Creemer wrote: Hi- I'm totally stumped configuring my TDM400P with one FXS and one FXO module. Before I got the FXO module, I used to have an X101P, and everything was working very well. Now * doesn't seem to recognize the FXO channel. I've searched the wiki and the list archives. Stock Debian 3.0 stable installation. Any advice? Thanks. I do not have an TDM400P, but read reports about it in this very list. Try replacing channel = 2 to 3 in zapata conf. The order of the modules seems to be relevant... -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Compact PCI platform
Hi, On Wed, 2004-05-19 at 18:07, Kyle Hagan wrote: I had considered trying this but from what I have read flash drive have a 1million read write life expetancy? If you were to use one of these as your harddrive would it not wear out pretty quick? Or am I wrong? Kyle One idea is to have a linux/asterisk version on the flash drive that boots and load everything into memory, with ramdisk et all. Similar to a linux/asterisk bootable from CD. But with the flash disk you can configure asterisk without the need to burn another CD or use a flacky floppy... The flash disk is only read when the machine boots. You can write to it for configuration data... 1 million times will last much more than a regular hard disk this way. But you will still need a hard disk for voicemails... -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What's in ${EXTEN} ? Why does voicemail prompt for an extension?
Hi,
On Fri, 2004-05-14 at 15:47, Paul Mahler wrote:
Why does voicemail prompt me for an extension instead of just asking my
password?
[voice-mail]
exten = 99,1,VoicemailMain([EMAIL PROTECTED])
exten = 99,2,Hangup
super snip
exten = 98,1,SayDigits(${EXTEN})
You might want to try with ${CALLERIDNUM} instead of ${EXTEN}
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Re: [Asterisk-Users] Re: Digits in a different language...
On Thu, 2004-05-06 at 18:15, Carlos Chavez wrote: On Thu, 06 May 2004 18:45:15 +0100, Fran Boon wrote Carlos Chavez wrote: All numbers like 10,20,30,40,50,60,70,80,90 and 100 have this problem. They all change when you have another number after. The only exception is the 1000 sound which does not change. For example: diez dieci (10) veinte veinti (20) From 30 to 90 you have to and an y (and) to the number. Some sounds like oh.gsm I simply recorded as cero (zero). Hi Carlos, I'm testing asterisk cvs-head as of yesterday. And the say_digits in spanish seems to works fine. You have to add the proper .gsm sounds: 20 thru 29, cien, 100, 200, 300, 400, 500, 600, 700, 800, 900, mil, millon, millones, y You have different sounds for 100: 100.gsm (ciento) and cien.gsm (cien) Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Flash panel
- Original Message - From: Altus Snyman [EMAIL PROTECTED] Good day all Did someone get the new ver0.5 flash panel working Is it suppose not to show who the caller is calling,like on ver0.2? And how do I change the language Thanks Altus Hi Altus, There is a mailing list for the flash operator panel. Please use that mailing list for discussing the application. You can subscribe by sending an empty email to: [EMAIL PROTECTED] The panel works if its properly configured. You can translate or change the text information by editing the op_server.pl. Good luck, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Disconnects
Hi Matt, Increase your busycount to 6 or 7. I had that problem also with an X100P, and it went away increasing the busycount parameter. On Mon, 2004-04-19 at 20:28, Matt Riddell wrote: I am getting random disconnects about 5-10 times a day. The logs show nothing except that the call was hung up. The calls are from X100P-*-digium T1 card-carrier access channel bank II-analogue phone. It is happening to all users. Is it possible that this is coming from busydetect=yes? Does busydetect detect cadences etc for the hangup frequencies? I have busycount=3... Any ideas? Any more information I could provide? Kind regards, Matt Riddell -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash Operator Panel new version and Mailing List
Hi All, Version .04 of the Flash Operator Panel is now available. Someone donated a domain name for the project (thanks turcko!), so it is now available on http://www.asternic.org I have set up a mailing list for the application. So please post your comments, suggestions, bug reports and problems there and not in Asterisk-Users. You can subscribe to the mailing list sending an empty email to [EMAIL PROTECTED] The new version has configurable buttons. You can have more that a hundred buttons on the screen. Flash Operator Panel displays information about your Asterisk PBX activity in real time via a standard web browser with Flash plugin. You can see at a glance: * What extensions are busy, ringing or available * Who is talking and to whom (clid, context, priority) * SIP registration status and reachability * Number of users waiting on Queues You can perform these actions: * Hang-up a channel * Transfer a call leg via dragdrop Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P FXO PCI Card
On Sat, 2004-04-10 at 15:50, Thomas Gallaway wrote: I run 4 X100P's in our asterisk box. Just make sure you give each card it's own IRQ. Paul, Is the own IRQ per card a strict rule ? Becasue a I have a X100P + TDM400P on a SMP PIII box, the X100P is sharing IRQ 11 with usb-uhci and no problems so far. Note that there is no USB devices attached to the box just the host driver loaded as a module. Maybe I am crazy, maybe not, but I have four X100P sharing the same interrupt. It works ok. Just a little echo sometimes.. Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
My apologies to the list members, I sent the mail by mistake to all of you, while my intention was to send it to Matt Ridell only. I also made a typo in the naming convention for IAX2, you have to remove the slash after IAX2. If you have problems/questions/bug reports with the operator panel, please send them to me directly! I wont release the .fla source for now, maybe in the future. New versions of the application will be posted in http://sip.house.com.ar/operator , I'm cleaning some bugs in the server and in the flash applet also. Thanks, - Original Message - From: Nicolas Gudino [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel Hi Matt, I modify the server to accept IAX2 channels (I think). Can you try it out? You have to name them like IAX2/[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
Hi Eric, - Original Message - From: Eric Wieling [EMAIL PROTECTED] Sent: Friday, April 02, 2004 11:17 AM Subject: Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel Being able to have more buttons as well as changing the button size would be useful. What screen resolutions do you use, how many buttons do you need? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] xml output from * ?
Hi, On Thu, 2004-04-01 at 15:37, John Todd wrote: At 9:35 AM -0500 3/31/04, [EMAIL PROTECTED] wrote: Hi Yawl, I took delivery this morning of a used BetaBrite LED display sign which I promptly set about playing with. Having found a windows app that grabs XML headline files from places like Slashdot and CNN as well as stocks etc I had an idea. What if I could get it to display stats from *? Things like call volume, queue stats, message waiting info. Add my voice to the me too chorus, though I don't have the time or skills to write it either. This would almost certainly be an external application (not in Asterisk) since the manager interface could provide the relevant information. There are Perl modules for the BetaBrite, I think... dig around. You can look at the op_server.pl I wrote. It connects to asterisk manager port, perform some magic and outputs xml to flash clients. It might give you ideas on how to implement the betabrite interface. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
Hi Tony, On Fri, 2004-04-02 at 14:13, Tony Buser wrote: We're having a problem with transfering calls. Our channels are not the same as the extensions. We use words instead of numbers. So our config looks like this: SIP/HRUTTER,1,81101 Hildegard SIP/JFOLEY-GS, 2,81103 Jerry Consequently when I drag and drop to transfer a call to Jerry, it fails because it tries to transfer to an extension called JFOLEY-GS, but his extension is really 81103. I will try to take care of that, my asterisk universe is very limited, I did not think about other naming conventions and uses for the different types of channels. Btw, might want to make the code be a little more forgiving, we could only get it to recognize the channels when we made the names in all capital letters (SIP/HRUTTER). Version .03 is on the website, case insenstive and more channel types supported. I looked through your code to see if I could make some changes, unfortunatly I can't speak Italian! :) Me neither! I speak spanish..LOL. -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCE: Flash Operator Panel
Hi, On Fri, 2004-04-02 at 16:09, Tony Buser wrote: by the way, when I start up op_server.pl I get the following, even though everything appears to work ok. Use of uninitialized value in transliteration (tr///) at ./op_server.pl line 67, CONFIG line 35. Use of uninitialized value in string at ./op_server.pl line 68, CONFIG line 35. Try removing line 35 on your op_server.cfg, maybe its a blank line and the server does not handle that gracefuly. Its not harmfull anyways. -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ANNOUNCE: Flash Operator Panel
http://sip.house.com.ar/operator Its a server/client combo that displays the status of your Asterisk PBX in a web browser in real time. You can also perform some actions. Hang-up channels and Transfers via drag and drop. The difference with other similar tools is that it displays status in real time (no refreshing necessary), and its graphically appealing. It's a work in progress... so expect some bugs. I appreciate any feedback you can give me. Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie....
Hi, On Wed, 2004-03-31 at 16:00, Hall, Eric M. wrote: I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. You can certainly use Asterisk without Digium hardware. But some applications will not work out of the box, like music on hold and meetme. For them to work you may need to compile ztdummy (uncomment the appropiate line in zaptel Makefile), and make sure that your sip clients transmit silence. If you are running RedHat or Fedora, start asterisk with LD_ASSUME_KERNEL=2.4.1 Good luck, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pre-paid (new to asterisk, pls don't shoot on me)
Hi, On this subject - has anybody managed to implement a method of warning the caller that their call will expire? I've Two questions; Has anybody successfully implemented this, either by way of source changes or by using the T extension (possibly something obvious I've missed?) I made a patch to play a tone before absolutetimeout. Its ugly but it works, at least for me. http://bugs.digium.com/bug_view_page.php?bug_id=773 Best regards, Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Opinion poll: best SIP phones for asterisk?
Hi, As I'm doing this, I'm considering installing an asterisk box at my office (about 6-10 different phone stations) and would like to get opinions on the best quality and/or most well-supported SIP hard phones and SIP soft phone clients. I had great luck with sipura spa-2000 adapters. They can make 3 way conferences and supervised and blind transfers by themselves. My advise is, whatever you choose, try to use the same brand of phones. Don't mix sipuras with grandstreams and ciscos. Good luck, Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes
Hi Jan,
Try this:
exten = _3XX,1,SetVar(FAXFILE=/tmp/faxfor-${EXTEN}-${TIMESTAMP}.tif)
exten = _3XX,2,rxfax(${FAXFILE})
Good luck,
- Original Message -
From: Jan Baumann [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, March 28, 2004 7:09 AM
Subject: [Asterisk-Users] RxFax/spandsp: file-naming of received faxes
after successfully having installed RxFax/SpanDSP and some promising tests
(great piece of software, Steve!) I wonder if it is possible to avoid
overwriting the same tiff file over and over again.
Browsing the sourcecode of app_rxfax.c I found a magic '%d' flag being
parsed
out from the argument of rxfax(), but didn't manage to make that work.
extensions.conf:
exten = _3XX,1,rxfax(/tmp/faxfor-${EXTEN}-%d.tif)
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Re: [Asterisk-Users] SoftFAX/spandsp
Hi Eric, I was all day trying and came up with this: gs -q -sDEVICE=tiffg3 -sPAPERSIZE=a4 -r204x196 \ -dNOPAUSE -sOutputFile=$TIFFILE -- $PSFILE I'm using a modified version of salsafax/sambafax to enable a print2fax option for windows/linux clients. You add a printer to cups and share it via Samba. Then, you append a line with the fax number in the file you want to be faxed Fax-Nr 3433 and print it to the network printer from any application. The scripts extracts the number and then generates a call file for asterisk. Some ps files cannot be extracted, so I used an OCR application (gocr) to extract the text, maybe its overkill, but it works most of the time (here we send less than ten faxes a day, so its no problem for us). I will clean up the scripts and post them for others to use. Good luck, On Thu, 2004-03-25 at 21:19, Eric Wieling wrote: On Thu, 2004-03-25 at 09:33, Steve Underwood wrote: exten = 5678,1,txfax(/tmp/testfax.tif|caller) There are a zillion fax and tiff formats. I'm trying to figure out what output format I should tell GhostScript to use. Any suggestions on which format to try? These are the formats GhostScript can output: faxg3 faxg32d faxg4 tiff12nc tiff24nc tiffcrle tiffg3 tiffg32d tiffg4 tifflzw tiffpack -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura line 1 outgoing voice problem?
Maybe this helps. I have 4 sipuras on the same network as Asterisk. I had to make sure each line on the sipura uses a different sip port: 5060/5061 on the first one, 5062/5063 on the second, and so on. Best regards, - Original Message - From: Matt McIntyre To: [EMAIL PROTECTED] Sent: Tuesday, March 23, 2004 6:59 PM Subject: RE: [Asterisk-Users] Sipura line 1 outgoing voice problem? I am experiencing this same problem and was wondering if anyone has come to a resolution. I have contacted Sipura but have not heard any response yet and am having trouble determining for sure whether the problem resides with Asterisk or the Sipura. As I have noticed that there are many users on the list who use the Sipura unit without this problem (and even a fellow with one unit that worked and one that did) I think the Sipura must be suspect. __ Back in January I started having a problem with my Sipura (and there was at least one other on the list with the same problem) that if I answer an incoming call (via X100P) on line 1 of my Sipura, the caller cannot hear any voice from the internal extension. If the internal user puts the external user on hold (via flash hook) and returns, both directions of audio are fine. Line 2 never has had this problem. For the meantime, I switched the internal phones so that my wife's favorite phone is line 2 and I told her to not pick up with line 1. Not a very permanent solution :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] X100P Tone-based Supervisory Disconnect ?
- Original Message - From: Gelson Dias Santos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Does it mean * supports tome based disconnect? How can I turn it ok? That what my original question (i´m the original poster). Try with: busydetect=yes busycount=7 in zapata.conf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Use of Alert_Info with C7960?
- Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk-a-users-list [EMAIL PROTECTED] Subject: [Asterisk-Users] Use of Alert_Info with C7960? Using Asterisk CVS-03/20/04-11:54:56 with 7960 v6.2 and playing around with distinctive ringing, trying to make it work. Extensions.conf looks like: exten = 3010,1,SetVar(ALERT_INFO=3) ; selects Ringer exten = 3010,2,Dial(SIP/3010,15) exten = 3010,3,Voicemail2(u3010) exten = 3010,102,Voicemail2(b3010) exten = 3010,103,Hangup On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old Style and Synth Low. The first three choices produce different ringing sounds when selected from the display. I expected Alert_Info=3 to cause the C7960 to ring with the Old Style ringer, but it doesn't and setting it to 2 or 3 doesn't make any difference. Am I doing something wrong? Rich A search on the mailing list returned this: http://lists.digium.com/pipermail/asterisk-users/2004-February/036747.html Try using: exten = 555,1,SetVar(ALERT_INFO=Bellcore-dr3)Best regards,Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Use of Alert_Info with C7960?
- Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, March 20, 2004 8:55 PM Subject: Re: [Asterisk-Users] Use of Alert_Info with C7960? On the phone, Settings/Ring Type indicates: Chirp 1, Chirp 2, Old Style and Synth Low. The first three choices produce different ringing sounds when selected from the display. I expected Alert_Info=3 to cause the C7960 to ring with the Old Style ringer, but it doesn't and setting it to 2 or 3 doesn't make any difference. Am I doing something wrong? A search on the mailing list returned this: http://lists.digium.com/pipermail/asterisk-users/2004-February/036747.html Try using: exten = 555,1,SetVar(ALERT_INFO=Bellcore-dr3)Best regards,Nicolas The wiki indicates Alert_Info can be set to a number, and implies that number is the ringer type listed on the phone. Is there a way to select one of the internal ringer types via Alert_Info? Hi Rich, The different ring tones are features of the sip phone/adapter. I dont have any Ciscos, but I do have the Sipura SPA-2000. I'm using ALERT_INFO to set distinctive rings and it works great. But the name of the ringtone is different from the one I quoted for the Cisco: exten = 12,2,SetVar(ALERT_INFO=Bellcore-r3) Is the phone/adapter job to interpret the alert info and play the acording ring tone. If the phone expects Bellcore-dr3, you should send that. Best regards, Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Re: Limit on call in minuttes.
Hi Hans, http://bugs.digium.com/bug_view_page.php?bug_id=773 This patch plays a tone 40,30,20 and 10 seconds before absolutetimeout. -- Nicolas Gudino Buenos Aires - Argentina - Original Message - From: Hans-Henrik Andresen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, March 07, 2004 3:02 PM Subject: [Asterisk-Users] Re: Re: Limit on call in minuttes. Hi, This isn't quit good :( The caller have the message played, but the called person are cut off without any warning.. I hoped to be warned, like In 1 minnute the line will be disconected, or just som beep beep /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and distinctive ring
On Fri, 2004-03-05 at 21:04, Matt McIntyre wrote: Has anyone implemented distinctive ring for SIP devices in Asterisk? My searches revealed that there was a patch created at one time but I can't tell if it was accepted or not. Basically I have a Sipura analog adapter that I would like to have ring differently for internal calls vs external calls. Thanks guys, Matt Hi Matt, Try with: exten = 1000,1,SetVar(ALERT_INFO=Bellcore-r3) -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does it exist - DNS TX record?
- Original Message - From: Chris Lee [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, March 02, 2004 6:42 AM Subject: [Asterisk-Users] Does it exist - DNS TX record? When handed a URL type address for telephony, is there a DNS TX record (like MX but for telephone/Video) that could be looked up for an address to use to connect the call? I would like to have a gateway server (probably *) that anyone who knows the email address of a member of staff can use to connect to them with. If the details of this server were in my DNS then anyone trying to call someone at cybericom.co.uk could find the server to make the connection with. Look here: http://www.voip-info.org/wiki-DNS+SRV ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to create channem of type 'Zap'
On Mon, 2004-02-23 at 17:10, Wim Venneman wrote: Can anyone help me, (after a two day search, also on the mailing list) I have the following situation: Asterisk works fine, until I added a FXO card. (Digium) When I tried to call to the pstn I have the following error Executing Dial(SIP/Phone2-fc49, Zap/1/2355) in new stack [channels] language=en context=incoming signalling=fxs_ks usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 pickupgroup=1 immediate=yes musiconhold=default channel = 1 ^^^ is this a typo? If not, the channel = 1 should go on a line of its own. -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem
Hi, I had the exact same problem, and it was caused by my crappy ADSL connection. I had great download and upload speeds too, but inspecting it closer, there was a great deal of lost packets. The problem went away when I changed my ADSL provider. - Original Message - From: yair hakak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 19, 2004 4:54 AM Subject: [Asterisk-Users] help a poor newbie out with SIP choppy one-way problem Hello all, i have a one-way choppy sound problem that i can't fix... here are the relevant points 1. i am running 18.2.04 CVS on rhl 9.0 on a hosted server with a wide pipe up/down with no hardware, just SIP connections and voicepulse for outgoing IAX calls. 2. conecting to * with SJPhone (SIP) on a windows box that gets 1.5MB down and about 100K upload in speed tests (ADSL), so i'm pretty sure client bandwidth is not a problem either. the client can ping the server at 180-200ms as well. I've also tried x-lite and gotten the same issues. sip clients register fine, and i can hear incoming audio fine, but on the other end it is completely garbled. It is not an IAX problem; if i leave voicemail from the SIP client on * and try to pick it up it is garbled, but the voicemail prompts are crystal clear. there was a thread about this at the beginning of january - the only solution that came up was to sweep the windows box for worms - which i did, and i have no worms. if anyone who had the problem then has answers, or anyone else, i would be most grateful. thanks, yair _ Protect your PC - get McAfee.com VirusScan Online http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dtmf recording record and playback
You can use AGI, the example below uses asterisk-perl:
---
#!/usr/bin/perl -w
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
$AGI-setcallback(\mycallback);
$number = $AGI-get_data(input-number, 1, 8);
$AGI-say_number($number);
exit 0;
sub mycallback {
my ($returncode) = @_;
print STDERR MYCALLBACK: User Hungup ($returncode)\n;
exit($returncode);
}
- Original Message -
From: Ed Devine
To: [EMAIL PROTECTED]
Sent: Thursday, February 19, 2004 1:54 PM
Subject: [Asterisk-Users] dtmf recording record and playback
I want to be able to record dtmf digits and then play them back as a voice
file (i.e. your enter your telephone number into a file, and asterisk reads
it back later as a voice file).
The application is similar to voicemail applications, but would need to be
able to parse the digits (i.e. 33 would be parsed as thrity three, rather
than three, three, etc...).
Has anyone got an idea how this could be implemented.
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RE: [Asterisk-Users] agi scripting in perl - dealiing withunexpected disconnects gracefully / spurious DTMF
On Thu, 2004-02-19 at 19:00, Tim Petlock wrote: I need to do something like this because I've timed calls with a stopwatch and can't figure out why the records going into the CDR table are 20 seconds longer (or more) than the actual call time. I understand that the actual call time includes the time spent entering and validating data but I've sat and timed it with a stopwatch and the CDR is always longer than reality. -Tim Hi Tim, In my case, the CDRs are longer because asterisk last inbetween 5 an 10 seconds to detect the hangup. You should take that into account. -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls with incoming distinctive ring
Look into bugs.digium.com. I think there is a patch for doing what you want. - Original Message - From: Scott Bennett [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, January 19, 2004 11:01 PM Subject: RE: [Asterisk-Users] Calls with incoming distinctive ring So am I to assume this is not possible? Can someone let me know one way or another, or just at least flame me for asking? Hello List I have searched the lists, the wiki and the handbook and see how to use distinctive ring inside however I can't find incoming. I have 1 x100p and 2 phone numbers, My Voice calls are normal ring, my Fax are short short long. How do I tell * to route the call to an extension based on the ring candance? Is it possible? Right now it seems when the x100p sees the short short long it locks up and refuses to answer the line again. Thanks For Any Help You Can Provide! Scott ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] R2 support
On Mon, 2004-01-19 at 18:38, Olle E. Johansson wrote: LQ (Asterisk) wrote: Hi guys, I was reading that Steve Underwood is working on Asterisk R2 signalling support, and has the 95% of the work done. What is R2? I'm curious. A type of signaling for E1 lines. -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ultra-cheap asterisk box
On Thu, 2004-01-15 at 14:31, Chris Albertson wrote: I'm looking to do about the same thing, build very low cost systems. (I'm looking at putting Asterisk at some non-profit organizations.) but one thing you can't make a compromise on is reliabilty. It has to work and keep working for years to come. I was able to keep the price of a new PC to about $300 ad still use an ASUS mainboard and an AMD XP2600+ The trick is to add absolutly nothing not needed. No floppy, no CDROM so you can run off a 200W P/S. Next I'll experiment with a notebook sized IDE disk drives and to see if _underclocking_ the CPU reduces it's power comsumption enough that we can save one fan. I'm also looking at this. I was thinking on a system without a hard drive, booting from a pendrive or flashdive. I want to avoid moving parts, they always break or get dirty and are noisy. If there are other people working on this, we might join efforts and work together and came up with a small linux version with asterisk included, that can boot from a pendrive or a cdrom. -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP and AGI crash...
On Tue, 2004-01-13 at 13:55, Tristan 'Minty' Colgate wrote: Hi, I'm trying to use the say-ani agi asterisk-perl script and am experiencing crashes, I am also experienceing problems with the test-agi scripts shipped with asterisk. Are you running RedHat 9? If you do, try with this line before launching asterisk (with stock redhat 9 kernels): export LD_ASSUME_KERNEL=2.4.1 -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linux Sip UAs
On Mon, 2004-01-12 at 12:23, Maciek Kaminski wrote: Hi, What linux SIP UAs do You successfully use with Asterisk? Maciej Kaminski kphone work ok, but its very basic. http://www.wirlab.net/kphone/ -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AbsoluteTimeout Users Messages
Andy Powell wrote: Nicolas, I'd appreciate a copy of this if possible... got a url where I can grab it? Thanks You can grab a copy from the bugtracker: http://bugs.digium.com/bug_view_page.php?bug_id=773 I've already sent the disclaimer to Digium.. Best regards, ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AbsoluteTimeout Users Messages
Andy Powell wrote: I'd be nice to be able to play a tone (or message) at AbsoluteTimeout - N where N is a number os seconds before the cut-off... a bit like pay phones (used?) to do... I have implemented an 'horrible' patch that sort of works. I'm not very good at C, and I'm new to asterisk. It makes a tone at 40, 30, 20 and 10 second before absolute-timeout. I can provide you with the patch, but its really really ugly, with lots of if/endifs. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one way choppy sound problem !
I still have the problem, but I have noticed one interesting fact. I have choppy sound from SIP to PSTN, but the voicemail prompts sound great (asterisk generated sounds are working well)... I will keep trying and keep you informed. On Mon, 2004-01-05 at 13:22, WipeOut wrote: Michael Van Donselaar wrote: I have the same choppy sound problem on my server, my card is not sharing an interrupt and I am using G711 which is not hittng the P2 400 at all.. It seems there is a gremlin.. :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one way choppy sound problem !
I have a similar problem, with GS phones, X-Lite or Kphone. I tried all the codecs with the same result. Choppy sound in the direction SIP-Phone - pstn, but crystal clear sound the other way around. The only difference in my case is that I have two asterisks servers connected together via IAX2, the PSTN call is received in one asterisk, while the sip phones are in the other asterisk. Ex: pstn - * --iax2-- * -sip phone (GS, Xlite or Kphone) If I use an Xlite in the same asterisk as the pstn line, the sound is perfect in both ways. But when I answer the call in the second asterisk, the sound from the sip phone to pstn is choppy, with or without silence detection, and the sound from pstn to sip phone is perfect. The asterisk server with the pstn line is an old pentium 133, maybe thats the problem, I will try with a better machine and see how it goes. On Fri, 2004-01-02 at 06:23, Dawid Mielnik wrote: Hi all, I have my asterisk setup as following: IP 2 x E1 x-lite --- Asterisk --- PSTN When I place a call from x-lite to PSTN, the quality of the sound in the direction x-lite - PSTN is very bad. That is, the voice of the x-lite user, heard by the PSTN user is choppy and makes communication not very pleasant. The sound is choppy as if bits of data were lost. The strange thing is that the x-lite user hears the PSTN user fine ! In x-lite, I have swithed off sience detection (transmit silence - yes), this has improved the sound quality but did not eliminated the problem. I have fed a countinious sound into the microphone and still got chops in the sound. I have also tried changing the codecs gsm, alaw, ulaw - but I get the same problem with all of them. Maybe the problem lies somewhere in audio buffering settings on x-lite ? Has anyone ever had this sort of problem and managed to deal with it ? I would greatly appreciate your help ! Best regards, Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * crash when forward voicemail --Nicolas Gudino
Well, Eric and James have answered already. Personally, I use redhat (will upgrade to fedora soon), but using an unmodified kernel.org kernel compiled from source. Best regards, On Thu, 2004-01-01 at 15:25, JR Richardson wrote: Hey Nicolas, That did it. I ran that export command you suggested, then launched *, everything worked fine. I'm still looking for info on what that command actually does. Can you shed some light please? Thanks. JR Did you try with this line before launching asterisk (with stock redhat 9 kernels): export LD_ASSUME_KERNEL=2.4.1 Best regards, -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one way choppy sound problem !
Hi Steven, On Fri, 2004-01-02 at 14:55, Steven Critchfield wrote: What is the ping times between your 2 asterisk servers? In the archive I have documented before that IAX jitter buffer sometimes has problems on short ping time links. At the time we where on a private T1 with 4ms ping times. We re enabled our jitter buffer now that we are on a DSL connection and our ping time is between 56 and 70 ms. The ping time is about 35 ms, one server is on ADSL and the other a T1. I tried with different jitter buffer settings, but I really don't know how to tune them. I also tried disabling jitter buffers. I even tried using a sip call directly, without using IAX2 (so no jitter buffers apply, at least no iax jitter buffers), always with the same result: choppy sound from sip to pstn and perfect sound from pstn to sip. Using alaw or ulaw the choppiness is tolerable, with other codecs is prety bad. Are there any documents on how to tune jitter buffers? Thanks! -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * crash when forward voicemail message [problem solved]
Did you try with this line before launching asterisk (with stock redhat 9 kernels): export LD_ASSUME_KERNEL=2.4.1 Best regards, On Tue, 2003-12-30 at 20:07, JR Richardson wrote: Thanks for all your help Martin, Guys, This is a good find and hopefully could help someone else. I've been having a problem with forwarding voicemail from one mailbox to another. I ran down the sendmail and soundcard path and came up goose eggs. With intuitive guidance from Martin Pycko (Digium), I switched from Redhat 9 Kernel linux-2.4.20-8 to Redhat 8 Kernel linux-2.4.18-14 and it seemed to solve the problem I was having. There is still a little weirdness going on but the voicemail forward command is working. During a -dgc session, I ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: * crash when forward voicemail message [problem solved]
RedHat 9 and Fedora kernels have a new feature (not present in kernel.org): Native Posix Threads This brings all sort of problems to diferente applications. To override this new feature, you have to start your affected programs with that enviroment variable set. On Tue, 2003-12-30 at 21:43, JR Richardson wrote: -Original Message- No I didn't, I don't have a clue what that is or does. Please explain, I'll try it and let you know. Did you try with this line before launching asterisk (with stock redhat 9 kernels): export LD_ASSUME_KERNEL=2.4.1 -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] International calling forbidden?
Hello,
On Thu, 2003-12-18 at 13:06, Michael Graves wrote:
[outbound-analog-int'l]
; allowed to call interntional long distance numbers via PSTN
; dial 8 to signify overseas calling
exten = _8011,1,Dial(${PSTNOUTBOUND/${EXTEN},70)
exten = _8011,2,Macro(fastbusy)
The number I'm calling is 011 44 1223 721 000. What am I doing wrong?
The number you are calling (011 44 xxx) does not match the dialplan. You
have to remove the 8:
exten = _011,1,Dial(${PSTNOUTBOUND/${EXTEN},70)
exten = _011,2,Macro(fastbusy)
If you want to 8 signify overseas, as the comentary line says,
you should dial 8 before 011, and remove one digit from the extension,
in order to not send that 8 to the PSTN.
exten = _8011,1,Dial(${PSTNOUTBOUND/${EXTEN:1},70)
exten = _8011,2,Macro(fastbusy)
All of this will work if you are including this context in
the proper place. Best regards,
--
Nicolas Gudino [EMAIL PROTECTED]
House Internet S.R.L.
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Re: [Asterisk-Users] asterisk phone card application with agi
Use the language you like/know more. I have developed a calling card application with AGI scripts in perl (with the help of asterisk-perl from http://asterisk.gnuinter.net) , web admin scripts in PHP, and MySQL backend. Good luck, On Wed, 2003-12-17 at 09:47, arun parajuli wrote: hey i want to implement phone card application based on PIN. for this i am planning to use the AGI. which programming language ( c , python, java .etc) should i use? i mean which one is effective. please suggest me. _ The new MSN 8: smart spam protection and 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (no subject)
I'm not a GPL expert, so I have a few questions: Does an AGI script needs to be distributed in source form? Maybe this application/script is using Asterisk unmodified. They can sell just their AGI scripts and provide only asterisk with full source? - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 09, 2003 3:48 AM Subject: Re: [Asterisk-Users] (no subject) Well if it links to asterisk and or used any of its code as a base it can't be sold without a comercial lic. for asterisk. Thats my understanding of the GPL. If its sold then all the source has to go along with it right? bkw On Tue, 9 Dec 2003, Adam Hart wrote: Is there a company website? or just a free yahoo email address? - Original Message - From: Kita B. Ndara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, December 09, 2003 4:01 PM Subject: [Asterisk-Users] (no subject) Hi, Our firm has developed two applications that I thought might be of interest to members of this list as both run over Asterisk: The first is a calling card application that covers needs in that area: scratch number generation, call termination via least-cost route (i.e. multiple termination providers), etc. We have tested this with voicepulse as our termination provider and it works great. The second is a call centre system: Call queueing, distribution, real-time reporting, statistics. Backend database is PostgreSQL (with pgcrypto module) for both applications, and in keeping with the Asterisk spirit, call origin/destination is h/w and software independent. If anybody is interested in these, please contact me off-list and I'll be happy to discuss these with you. Thanks B. BT Yahoo! Broadband - Save £80 when you order online today. Hurry! Offer ends 21st December 2003. The way the internet was meant to be. http://uk.rd.yahoo.com/evt=21064/*http://btyahoo.yahoo.co.uk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: FXO cards
- Original Message - From: Michael Rowley [EMAIL PROTECTED] So, the docs say no more than 2 x100p cards sane, has anyone done it? put 5 or 6 in one box? I'm using 4 of them, it works. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call pickup and SIP phones
Hi List, I have two Cisco ATA, one of them with two phones attached, and the other with just one phone. The ATA with two phones is behind a NAT, and Asterisk and the other ATA have public IP addresses. I can place and receive and blind transfer calls between them all. (Sometimes I loose registration from the ATA behind the NAT, but I think I have to upgrade to the latest firmware in the ATA) Now I'm trying to setup call pickup. I added the lines: pickupgroup=1 callgroup=1 to every entry in sip.conf, but when I try to pickup a call dialing *8 or *8# from the idle phone , nothing happens. I'm using CVS version CVS-10/10/03-19:24:38. In the console nothing shows up either. Do I have to upgrade to a more recent CVS version? Do I need to enter more parameters or configurations in other places? Does anyone have call pickup between sip phones working? If so, which version are you using? Thanks!! -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI questions..
Hi, I use asterisk-perl and when I execute Dial from my script, it stops procesing it and I cannot perform cleanups or post call operations in that script. It would be very nice to take the control back from the script.. Finally, If I execute a call from within an AGI script, will the script continue processing when the call is hung up or terminated or would I have to use another AGI on the h extension to process post call operations? Good question. I can't answer. This is an important question I need answered for my system.. -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI questions..
Hi, I'm trying to use the h extension for postprocessing, but for now I couldnt do it, it runs, but lacks debuging output; well, to tell you the thruth, I didnt try to hard to make it work from the h extension. What I do is this: the script is for controling prepaid accounts. I do the post-call processing at the beginning of the script (that is, calculate the cost of the pending processing calls and substract the amount to the account, *before* placing a new call). So, the last call is not processed until the user tries to make another one. It works perfectly well in this way, but I think its cleaner to calculate the cost after the user hangs ups. Hi, I use asterisk-perl and when I execute Dial from my script, it stops procesing it and I cannot perform cleanups or post call operations in that script. It would be very nice to take the control back from the script.. So how do you do your post call operations?? do you use another AGI script on the h extension? Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with CDR dst when executing Dial from 's' extension
Hi list, I have a little problem that I cannot solve myself, being new to Asterisk. Maybe someone can help me out. I'm executing a Dial command in an AGI script launched from the 's' extension. It works great, but the CDR shows 's' as the destination. I need a CDR record with the number passed to the dial command as dst in the CDR. I'm sure there is a proper way to handle this.. and I'm sure someone can help me figure it out. Thanks! Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI problem (crash) in RH9
Hi Ivar, Try putting this line before launching asterisk: export LD_ASSUME_KERNEL=2.4.1 Best regards, On Thu, 2003-10-16 at 06:48, var Ragnarsson wrote: Hi Every time I hangup on my AGI script Asterisk crashes if it is not running in console mode. (happens when using python and perl AGI scripts) I'm desparatly trying to get my employer to let me use Asterisk. So I must get this to work. I've posted about this before, I'm sorry, but I'm desperate. I'm running RedHat 9.0 (kernel 2.4.20-8 everything else updated) I'm using Netmeeting to test -- Nicolas Gudino [EMAIL PROTECTED] House Internet S.R.L. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI problem (crash)
Hi, I have the same problem, Im also running RH 9. But Im using SIP only with Cisco ATAs. There are reports of asterisk not doing well with RedHat because of the new threads handling in RH kernel. Maybe compiling a fresh rpm from kernel.org will solve the problem. Testing my AGI script (writen in perl) I had a dial command (not background), and I could cause asterisk to crash when hanging up when inside the script. But now I moved the dial command to the extensions file (setting a variable inside the script to pass to the dial command in the extension file) and the problem went away. Maybe it is because its difficult to hang up exactly during the execution of the script that runs and exits really fast now that it is not doing the dial. I will try to compile a fresh kernel and see if the problem persists, and post my results here. Regards, On Thu, 2003-10-16 at 06:48, var Ragnarsson wrote: Hi Every time I hangup on my AGI script Asterisk crashes if it is not running in console mode. (happens when using python and perl AGI scripts) I'm desparatly trying to get my employer to let me use Asterisk. So I must get this to work. I've posted about this before, I'm sorry, but I'm desperate. I'm running RedHat 9.0 (kernel 2.4.20-8 everything else updated) I'm using Netmeeting to test I use H.323 only. I've tried using chan_h323 and chan_oh323. (Output is from oh323) Newest zaptel libpri asterisk from CVS Has anyone had this problem? Can anyone confirm this failure on a similar system? (that is running the script and hanging up while the number is beeing read.) Can anyone test this on RedHat 8 please? Are there any log files that could give clues to what is happening? Should I post this on the dev mailing list? Following are outputs from the console, a sample script and my config files. ++ output from asterisk -vvv ++ [chan_oh323.so] = (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.007 OpenH323 Wrapper OpenH323 WrapperVersion 0.0alpha0 by inAccess Networks (www.inaccessnetworks.com) on Unix Linux (2.4.20-8-i686) at 2003/10/15 15:34:17.735 WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.12.0, PWlib v1.5.0 == Registered channel type 'OH323' (OpenH323 Channel Driver) == OpenH323 Channel Ready (v0.5.5) == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Executing Answer(H323:25128, ) in new stack PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. -- Executing AGI(H323:25128, agi-pytest2.py) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-pytest2.py -- Playing 'digits/1' -- Playing 'digits/hundred' PAsteriskSoundChannel::PAsteriskSoundChannel: Object deleted. PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted. PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted. PAsteriskSoundChannel::PAsteriskSoundChannel: Object deleted. PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted. PAsteriskAudioDelay::PAsteriskAudioDelay: Object deleted. 0:05.269 H323 Cleaner H323Connection ip$192.168.0.100:1712/25128 terminated. == Spawn extension (default, 147, 2) exited non-zero on 'H323:25128' -- Hungup 'H323:25128' Received 200 result=-1 ++ output from asterisk -vvvc ++ [chan_oh323.so] = (OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found 0:00.008 OpenH323 Wrapper OpenH323 WrapperVersion 0.0alpha0 by inAccess Networks (www.inaccessnetworks.com) on Unix Linux (2.4.20-8-i686) at 2003/10/15 15:35:11.096 WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.12.0, PWlib v1.5.0 == Registered channel type 'OH323' (OpenH323 Channel Driver) == OpenH323 Channel Ready (v0.5.5) == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Executing Answer(H323:25129, ) in new stack PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized. PAsteriskSoundChannel::PAsteriskSoundChannel: Object initialized. PAsteriskAudioDelay::PAsteriskAudioDelay: Object initialized.
[Asterisk-Users] Licensing G729
Hi List, I'm new to asterisk. I think it's great! I'm interested in terminating calls via a SIP provider. I want to know if I need to license G729 on asterisk in these scenarios: CISCO ATA186 - Asterisk - SIP Provider - PSTN or this one: CISCO ATA186 - Asterisk - CISCO ATA To my understanding, in the second case, if one of the ATA is behind NAT, I should set canreinvite=no, so the RTP channels would go through *, so I would have to license G729 in order to use this codec with the ATAs. Is this right? But if boths ATA have public IPs, and * issues a reinvite, can the ATAs negotiate G729 themselves, without needing it on * ? And in the first scenario, if the SIP provider supports G729 and the ATA has a public IP, do I need to license the codec in *? Thanks in advance, Nicolas Gudino Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream 102
Sorry about this off-topic question... I want to know if the second ethernet port on the Grandstream 102 phone works as a bridge to connect from there to a PC. Do I need two ethernet jacks to connect a phone and a PC, or this phone let me connect both with only one? Thanks in advance! Nicolas Gudino
