[Asterisk-Users] no voice on dialogic d300
I am trying to test my dialogic card with asterisk , ihave an E1 card in asterisk and onedialogic(d300) in anothermachine. Both are connected through a cross cable. Asterisk(digium E1)calls the dialogic(d300) card to its answer demo. The answer demo shows that the call is received but there is no voice. Can some one tell me the list of things to check or change. Like a-law,u-law for example. Thanks Regards Ayaz Gul Aga
RE: [Asterisk-Users] where to get adsi phones in europe ?
is there somebody who can help me with getting ADSI phones in Europe I' am a little bit desperated. I need such a phone to play with * and adsi features. But i don't find a vendor who produce or a distributor who distribute such phones in Europe. I have found this link in the ast-users-mailing list -- http://lktelecom.zoovy.com/product/HPT350 This is the Power Touch 350. But the delivery costs are more expencive than the product self! The telephone cost 49$ and the shipping would cost 79$! This is unacceptable. I didn't realise ADSI capable phones could be this cheap!! Can someone confirm that these really are ADSI phones, will work with asterisk, and are 'unlocked' etc. Also, does anyone know of anywhere is Australia with these type of phones? If not, if anyone in Australia is interested in purchasing some of these phones, perhaps I can place a bulk order to reduces the shipping costs for each of us. BTW, has anyone else used/seen these phones? Are they any good? Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] databases for billing
TWO THINGS CARLOS ! one, please turn off your html formatting. second, it was answered (and even can be seen in your posting..) cdr_mysql.conf which you will find in /etc/asterisk actually the original is cdr_mysql.conf.sample have read pls. On Mon, 23 Jun 2003 09:02:22 +0200 (CEST), carlos del mayor wrote: I'm only asking for some examples of cdr_mysql.conf, nobody has done anything with cdr and mysql? If you think is better another DB,,, tell me, please! thanks in advance carlos carlos del mayor [EMAIL PROTECTED] wrote: can you be more explicit, please? or give me some examples? please, i'm little lost! thanks a lot carlos Martin Pycko [EMAIL PROTECTED] wrote: cdr_mysql.conf On Fri, 20 Jun 2003, carlos del mayor wrote: hi I want to do a database to save the cdr with a billing finality. I've created the database in mysql (thanks for the table and all that!) but I'm not sure of how to 'connect' asterisk to that database in order to save there the cdr. Is the cdr_mysql.conf what I have to config? Or must I do a script, with the 'database' AGI commands? Any help would be so apreciated! Thanks a lot carlos - Yahoo! Sorteos Juega a la Loter¡a Primitiva sin salir de casa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - Ya! hoo! Sorteos Juega a la Loter¡a Primitiva sin salir de casa - Yahoo! Sorteos Juega a la Loter¡a Primitiva sin salir de casa . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] databases for billing
TWO THINGS,GARY! 1-Sorry for the html, now it's off 2-The mail you're talking about has arrived two minutes ago.I KNOW read, thank you. I only wanted to know if somebody was working with this, in order to simplify a litle my work (documentation and all that it's what i was looking for). Gary, I don't think your reaction is ok, people who are starting with new features (like me) only demand a litle of help and patience, only that. I know that we are a lot, and there are less 'guru' pepople than newbie one, but a mailing list is used to help. You can 'not help', I don't mind, but please don't send messages that can make newbie people feel this mailing list is not for them. Thanks for everybody that helps without desperating. And sorry if I make you desperate! Carlos --- Gary [EMAIL PROTECTED] escribió: TWO THINGS CARLOS ! one, please turn off your html formatting. second, it was answered (and even can be seen in your posting..) cdr_mysql.conf which you will find in /etc/asterisk actually the original is cdr_mysql.conf.sample have read pls. On Mon, 23 Jun 2003 09:02:22 +0200 (CEST), carlos del mayor wrote: I'm only asking for some examples of cdr_mysql.conf, nobody has done anything with cdr and mysql? If you think is better another DB,,, tell me, please! thanks in advance carlos carlos del mayor [EMAIL PROTECTED] wrote: can you be more explicit, please? or give me some examples? please, i'm little lost! thanks a lot carlos Martin Pycko [EMAIL PROTECTED] wrote: cdr_mysql.conf On Fri, 20 Jun 2003, carlos del mayor wrote: hi I want to do a database to save the cdr with a billing finality. I've created the database in mysql (thanks for the table and all that!) but I'm not sure of how to 'connect' asterisk to that database in order to save there the cdr. Is the cdr_mysql.conf what I have to config? Or must I do a script, with the 'database' AGI commands? Any help would be so apreciated! Thanks a lot carlos - Yahoo! Sorteos Juega a la Loter¡a Primitiva sin salir de casa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - Ya! hoo! Sorteos Juega a la Loter¡a Primitiva sin salir de casa - Yahoo! Sorteos Juega a la Loter¡a Primitiva sin salir de casa . ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Yahoo! Sorteos - http://loteria.yahoo.es Juega a la Lotería Primitiva sin salir de casa ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Module app_perl
On Tuesday, Jun 17, 2003, at 20:43 America/Los_Angeles, Anthony Minessale wrote: Here is a copy of the first release (comments appreciated) http://asterisk.650dialup.com Although I haven't had time to play with it: very neat! - ask -- http://www.askbjoernhansen.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help with pri configuration..
Hi all, can somebody help me with pri configuration? Here my zapata.conf: ; Zapata telephony interface ; ; Configuration file [channels] switchtype=euroisdn signalling=pri_cpe ;group=1 channel = 1-15,17-31 ;group=2 channel =32-46,48-62 ;group=3 channel = 63-77,79-93 ;group=4 channel = 94-108,110-124 And here my zaptel.conf: zaptel.conf [] 0 L:[ 1+ 0 1/ 18] *(0 / 227b)= . 10 0x0A span=1,0,0,ccs,hdb3 #,crc4 span=2,0,0,ccs,hdb3 #,crc4 span=3,0,0,ccs,hdb3 #,crc4 span=4,0,0,ccs,hdb3 #,crc4 bchan=1-15,,32-46,63-77,94-108 dchan=16,47,78,109 bchan=17-31,48-62,79-93,110-124 And here the messages after starting astersik: loadzone = fr defaultzone=us [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Registered channel 3, PRI Signalling signalling -- Registered channel 4, PRI Signalling signalling -- Registered channel 5, PRI Signalling signalling -- Registered channel 6, PRI Signalling signalling -- Registered channel 7, PRI Signalling signalling -- Registered channel 8, PRI Signalling signalling -- Registered channel 9, PRI Signalling signalling -- Registered channel 10, PRI Signalling signalling -- Registered channel 11, PRI Signalling signalling -- Registered channel 12, PRI Signalling signalling -- Registered channel 13, PRI Signalling signalling -- Registered channel 14, PRI Signalling signalling -- Registered channel 15, PRI Signalling signalling -- Registered channel 17, PRI Signalling signalling -- Registered channel 18, PRI Signalling signalling -- Registered channel 19, PRI Signalling signalling -- Registered channel 20, PRI Signalling signalling -- Registered channel 21, PRI Signalling signalling -- Registered channel 22, PRI Signalling signalling -- Registered channel 23, PRI Signalling signalling -- Registered channel 24, PRI Signalling signalling -- Registered channel 25, PRI Signalling signalling -- Registered channel 26, PRI Signalling signalling -- Registered channel 27, PRI Signalling signalling -- Registered channel 28, PRI Signalling signalling -- Registered channel 29, PRI Signalling signalling -- Registered channel 30, PRI Signalling signalling ERROR[1024]: File chan_zap.c, Line 4757 (mkintf): Signalling requested is PRI Signalling but line is in Unkn own signalling 896 signalling ERROR[1024]: File chan_zap.c, Line 6403 (load_module): Unable to register channel '1-15' WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! Whats wrong ? Thanks for help, Thomas. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Process multiple commands on dial out..
Hi,
How do I process multiple lines of the extention.conf on dial out before actually
connecting the call to the user??
Here is the problem..
I have an access number for cheap internationsl calls, This number has to be dialed
and then a DTMF string needs to be passd to the service for the number that is being
called in the destination country..
When I was using an analog line I could do it all in one string but now that I am
using ISDN it requires three lines in the extensions.conf file like this.. (As
suggested by Kapjod)
exten = _900.,1,Dial(CAPI/[msn here]:[access number])
exten = _900.,2,Wait(3)
exten = _900.,3,SendDTMF(${EXTEN:1})
The problem is that the call is connected to the user on the first line and so the
SendDTMF is never processed and so the international number is never dialed..
Anyone got an idea how I can do this??
Thanks..
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AW: [Asterisk-Users] help with pri configuration..
The problem before is solved. But now gives another problem ... == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) == Starting D-Channel on span 1 ERROR[1024]: File chan_zap.c, Line 5947 (start_pri): Unable to open D-channel 47 (Device or resource busy) ERROR[1024]: File chan_zap.c, Line 6682 (load_module): Unable to start D-channel on span 2 WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! All channels are registered successfully before. But then this error occur. I tried to deactivate following ports (spans) 1. second one but then the same message occure with the next dchannel and so on. Only the first one works. What's wrong now? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Montag, 23. Juni 2003 11:34 An: Asterisk User Betreff: [Asterisk-Users] help with pri configuration.. Hi all, can somebody help me with pri configuration? Here my zapata.conf: ; Zapata telephony interface ; ; Configuration file [channels] switchtype=euroisdn signalling=pri_cpe ;group=1 channel = 1-15,17-31 ;group=2 channel =32-46,48-62 ;group=3 channel = 63-77,79-93 ;group=4 channel = 94-108,110-124 And here my zaptel.conf: zaptel.conf [] 0 L:[ 1+ 0 1/ 18] *(0 / 227b)= . 10 0x0A span=1,0,0,ccs,hdb3 #,crc4 span=2,0,0,ccs,hdb3 #,crc4 span=3,0,0,ccs,hdb3 #,crc4 span=4,0,0,ccs,hdb3 #,crc4 bchan=1-15,,32-46,63-77,94-108 dchan=16,47,78,109 bchan=17-31,48-62,79-93,110-124 And here the messages after starting astersik: loadzone = fr defaultzone=us [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Registered channel 3, PRI Signalling signalling -- Registered channel 4, PRI Signalling signalling -- Registered channel 5, PRI Signalling signalling -- Registered channel 6, PRI Signalling signalling -- Registered channel 7, PRI Signalling signalling -- Registered channel 8, PRI Signalling signalling -- Registered channel 9, PRI Signalling signalling -- Registered channel 10, PRI Signalling signalling -- Registered channel 11, PRI Signalling signalling -- Registered channel 12, PRI Signalling signalling -- Registered channel 13, PRI Signalling signalling -- Registered channel 14, PRI Signalling signalling -- Registered channel 15, PRI Signalling signalling -- Registered channel 17, PRI Signalling signalling -- Registered channel 18, PRI Signalling signalling -- Registered channel 19, PRI Signalling signalling -- Registered channel 20, PRI Signalling signalling -- Registered channel 21, PRI Signalling signalling -- Registered channel 22, PRI Signalling signalling -- Registered channel 23, PRI Signalling signalling -- Registered channel 24, PRI Signalling signalling -- Registered channel 25, PRI Signalling signalling -- Registered channel 26, PRI Signalling signalling -- Registered channel 27, PRI Signalling signalling -- Registered channel 28, PRI Signalling signalling -- Registered channel 29, PRI Signalling signalling -- Registered channel 30, PRI Signalling signalling ERROR[1024]: File chan_zap.c, Line 4757 (mkintf): Signalling requested is PRI Signalling but line is in Unkn own signalling 896 signalling ERROR[1024]: File chan_zap.c, Line 6403 (load_module): Unable to register channel '1-15' WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! Whats wrong ? Thanks for help, Thomas. ___ 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] Budgetone + remote call pickup
Hi. I've found a problem when I pickup a remote sip phone with *8. There're both budgetones 102 and are both in the same group. When one sip phone is ringing, I can pickup the call from another sip phone, but the first one keeps playing a loud busy signal... that don't go away until I receive another call or go off hook and then on hook on the first phone. I think that could be a budgetone bug on BYE command, since the snom and the crisco works ok... But anyway I attached the log file (233 is the called, 225 is the one who pickups via *8). Anyone experienced that? Matteo. asterisk*CLI sip debug SIP Debugging Enabled -- Accepting unauthenticated call from 213.140.14.155, requested format = 4, actual format = 2 -- Executing AGI([EMAIL PROTECTED]:4569]/2, channel_lookup.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/channel_lookup.agi -- AGI Script channel_lookup.agi completed, returning 0 -- Executing Dial([EMAIL PROTECTED]:4569]/2, Sip/233|30|m) in new stack We're at 192.168.1.203 port 13938 Answering with preferred capability 4 10 headers, 7 lines Reliably Transmitting: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK41388b4a From: Guest IAX User sip:[EMAIL PROTECTED];tag=as3786d582 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Content-Type: application/sdp Content-Length: 135 v=0 o=root 19057 19057 IN IP4 192.168.1.203 s=session c=IN IP4 192.168.1.203 t=0 0 m=audio 13938 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (no NAT) to 192.168.1.243:5060 -- Called 233 -- Started music on hold, class 'default', on [EMAIL PROTECTED]:4569]/2 Sip read: LI SIP/2.0 100 trying Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK41388b4a From: Guest IAX User sip:[EMAIL PROTECTED];tag=as3786d582 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream SIP UA 1.0.3.60 Content-Length: 0 8 headers, 0 lines Sip read: LI SIP/2.0 180 ringing Via: SIP/2.0/UDP 192.168.1.203:5060;branch=z9hG4bK41388b4a From: Guest IAX User sip:[EMAIL PROTECTED];tag=as3786d582 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Grandstream SIP UA 1.0.3.60 Content-Length: 0 8 headers, 0 lines -- SIP/233-a2f4 is ringing 10 headers, 0 lines Sip read: LI INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235 From: sip:[EMAIL PROTECTED];tag=0da17af0-4d2a-1a7d-9744-548fad8a59bb To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1320 INVITE User-Agent: Grandstream SIP UA 1.0.3.60 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS Content-Type: application/sdp Content-Length: 314 v=0 o=225 0 0 IN IP4 192.168.1.235 s=- c=IN IP4 192.168.1.235 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 12 headers, 15 lines Using latest request as basis request Sending to 192.168.1.235 : 5060 (non-NAT) Capabilities: us - 4, them - 269, combined - 4 Non-codec capabilities: us - 1, them - 1, combined - 1 Reliably Transmitting (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.235 From: sip:[EMAIL PROTECTED];tag=0da17af0-4d2a-1a7d-9744-548fad8a59bb To: sip:[EMAIL PROTECTED];tag=as38509822 Call-ID: [EMAIL PROTECTED] CSeq: 1320 INVITE User-Agent: Asterisk PBX Contact: Proxy-Authenticate: Digest realm=asterisk, nonce=466877d9 Content-Length: 0 to 192.168.1.235:5060 Sip read: LI ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235 From: sip:[EMAIL PROTECTED];tag=0da17af0-4d2a-1a7d-9744-548fad8a59bb To: sip:[EMAIL PROTECTED];tag=as38509822 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 1320 ACK User-Agent: Grandstream SIP UA 1.0.3.60 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS Content-Length: 0 11 headers, 0 lines Sip read: LI INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.235 From: sip:[EMAIL PROTECTED];tag=1a7d0da1-548f-4d2a-59bb-9744a2d7ad8a To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Proxy-Authorization: DIGEST username=225, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=466877d9, response=0c894fd4b402750275650e18a138123e Call-ID: [EMAIL PROTECTED] CSeq: 1321 INVITE User-Agent: Grandstream SIP UA 1.0.3.60 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS Content-Type: application/sdp Content-Length: 314 v=0 o=225 0 0 IN IP4 192.168.1.235 s=- c=IN IP4 192.168.1.235 t=0 0 m=audio 5004 RTP/AVP 0 8 4 18 2 15 101 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:15 G728/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 13 headers, 15
Re: [Asterisk-Users] Billsec on CDR
malaysia and sri lanka toowould be greately appreciated :-)- Original Message -From: "Michael Labuschke" [EMAIL PROTECTED]To: [EMAIL PROTECTED]Sent: Saturday, June 21, 2003 11:48 PMSubject: Re: [Asterisk-Users] Billsec on CDR germany here :))) *** REPLY SEPARATOR *** On 21.06.2003 at 08:51 Stephen Davies wrote: On Fri, 20 Jun 2003, Tan Aks wrote: Isn't there any way to make callprogress work for people in Europe?What is it that is needed to make it work? I've done call progress for the UK. Patch to the -dev list by the end of the weekend. What country do you want? ! Steve ___ 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 --This mail sent through OmniBIS.com-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] no voice on dialogic d300
I am getting the calls through but there is no voice , help guys !!! - Original Message - From: Jordan Peterson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 23, 2003 11:14 AM Subject: Re: [Asterisk-Users] no voice on dialogic d300 This could be related to my not hearing voice through the phone over a voice modem as well, which could answer both our problems if we have a routine of things to check off. thanks On Sun, 2003-06-22 at 22:47, ayaz wrote: I am trying to test my dialogic card with asterisk , i have an E1 card in asterisk and one dialogic(d300) in another machine. Both are connected through a cross cable. Asterisk (digium E1) calls the dialogic(d300) card to its answer demo. The answer demo shows that the call is received but there is no voice. Can some one tell me the list of things to check or change. Like a-law,u-law for example. Thanks Regards Ayaz Gul Aga -- Jordan In a world without windows, who needs gates? ___ 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] Dialogic Proline 2v Supported?
Anyone know if the Dialogic/Intel Proline 2v supported by Asterisk? _ Help STOP SPAM with the new MSN 8 and get 2 months FREE* http://join.msn.com/?page=features/junkmail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P and Caller ID on Call Waiting
Hi there! I'm having a problem with TDM400P and Caller ID on Call Waiting. Normal Caller ID works quite well, but I can't get CIDCW to work (tested against Siemens phones). I hear the tone, but a message on the console appears telling that the phone doesn't support CIDCW; when it does (it's used every day against the telephone central, a Siemens EWSD). Any ideas what could be going on? Thanks a lot, Alberto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] codecs question ..
You need G723 CODEC to be supportted on your asterisk server. Best regards Lubo Dave Alan Caruana wrote: My system is an asterisk machine, with an E1 card (functioning) and forwarding calls to a remote SIP address .. when a call connects I am getting the following error : NOTICE[1240577216]: File rtp.c, Line 330 (ast_rtp_read): Unknown RTP codec 19 received can anybody tell me what this means ( how I may fix it ?) cheers Dave ps. the service i'm connecting to uses G723 ___ 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] Setting up the E100P
On Monday 23 June 2003 2:58 pm, Anton Yurchenko wrote: Hello, I have an E100P, and in the zaptel.conf I have: span=1,1,0,ccs,hdb4,crc4,yellow fxsks=1-10 delete the fxsks line and put: bchan=1-15,17-31 dchan=16 the light on the card is green( BTW what do all those states of the card that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or for the card?) in the asterisks` zapata.conf I have: [channels] context=default switchtype=euroisdn 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 callgroup=1 pickupgroup=1 immediate=no callerid=line1 238-20-31 channel = 1 callerid=line2 238-20-31 channel = 2 but Asterisk on startup reports that: WARNING[13326]: File chan_zap.c, Line 4173 (ss_thread): CallerID returned with rror on channel 'Zap/1-1' WARNING[14351]: File chan_zap.c, Line 4173 (ss_thread): CallerID returned with rror on channel 'Zap/2-1' and when I call the number that is supposed to terminate there I get busy signal. also in the zttool I see strange thing, in the bottom portion where there are RxA, RxB the 111 ( ones) sometimes change into 00, they go from up to down and if I enable all the channels then firs on the first column then on the second column, in the bottom half of the screen. while if I plug the same E1 into Cisco AS5300, whit this config( just exepts): isdn switch-type primary-net5 controller e1 ... clock source line primary pri-group timeslots 1-31 ... interface Serial0:15 isdn switch-type primary-net5 isdn incoming-voice modem 64 Framing is CRC4, Line Code is HDB3, Clock Source is Line ... on Cisco the swithtype is primary-net5 ( my guess its euroisdn ? ) anybody could guess what is the problem? The admin that runs the Cisco says that signalyng should be PRI , and there is an option for pri signaling in zapata.conf, but the zaptel,conf doesnt have it and so I`d get up with a mismatch, and zasterisk would not start. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] help with pri configuration..
Well how did you solve your previous problem then ? Martin On Mon, 23 Jun 2003, Thomas Haeger wrote: The problem before is solved. But now gives another problem ... == Registered channel type 'Zap' (Zapata Telephony Driver w/PRI) == Registered channel type 'Tor' (Zapata Telephony Driver w/PRI) == Starting D-Channel on span 1 ERROR[1024]: File chan_zap.c, Line 5947 (start_pri): Unable to open D-channel 47 (Device or resource busy) ERROR[1024]: File chan_zap.c, Line 6682 (load_module): Unable to start D-channel on span 2 WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! All channels are registered successfully before. But then this error occur. I tried to deactivate following ports (spans) 1. second one but then the same message occure with the next dchannel and so on. Only the first one works. What's wrong now? Thanks, Thomas. -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Thomas Haeger Gesendet: Montag, 23. Juni 2003 11:34 An: Asterisk User Betreff: [Asterisk-Users] help with pri configuration.. Hi all, can somebody help me with pri configuration? Here my zapata.conf: ; Zapata telephony interface ; ; Configuration file [channels] switchtype=euroisdn signalling=pri_cpe ;group=1 channel = 1-15,17-31 ;group=2 channel =32-46,48-62 ;group=3 channel = 63-77,79-93 ;group=4 channel = 94-108,110-124 And here my zaptel.conf: zaptel.conf [] 0 L:[ 1+ 0 1/ 18] *(0 / 227b)= . 10 0x0A span=1,0,0,ccs,hdb3 #,crc4 span=2,0,0,ccs,hdb3 #,crc4 span=3,0,0,ccs,hdb3 #,crc4 span=4,0,0,ccs,hdb3 #,crc4 bchan=1-15,,32-46,63-77,94-108 dchan=16,47,78,109 bchan=17-31,48-62,79-93,110-124 And here the messages after starting astersik: loadzone = fr defaultzone=us [chan_zap.so] = (Zapata Telephony w/PRI) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, PRI Signalling signalling -- Registered channel 2, PRI Signalling signalling -- Registered channel 3, PRI Signalling signalling -- Registered channel 4, PRI Signalling signalling -- Registered channel 5, PRI Signalling signalling -- Registered channel 6, PRI Signalling signalling -- Registered channel 7, PRI Signalling signalling -- Registered channel 8, PRI Signalling signalling -- Registered channel 9, PRI Signalling signalling -- Registered channel 10, PRI Signalling signalling -- Registered channel 11, PRI Signalling signalling -- Registered channel 12, PRI Signalling signalling -- Registered channel 13, PRI Signalling signalling -- Registered channel 14, PRI Signalling signalling -- Registered channel 15, PRI Signalling signalling -- Registered channel 17, PRI Signalling signalling -- Registered channel 18, PRI Signalling signalling -- Registered channel 19, PRI Signalling signalling -- Registered channel 20, PRI Signalling signalling -- Registered channel 21, PRI Signalling signalling -- Registered channel 22, PRI Signalling signalling -- Registered channel 23, PRI Signalling signalling -- Registered channel 24, PRI Signalling signalling -- Registered channel 25, PRI Signalling signalling -- Registered channel 26, PRI Signalling signalling -- Registered channel 27, PRI Signalling signalling -- Registered channel 28, PRI Signalling signalling -- Registered channel 29, PRI Signalling signalling -- Registered channel 30, PRI Signalling signalling ERROR[1024]: File chan_zap.c, Line 4757 (mkintf): Signalling requested is PRI Signalling but line is in Unkn own signalling 896 signalling ERROR[1024]: File chan_zap.c, Line 6403 (load_module): Unable to register channel '1-15' WARNING[1024]: File loader.c, Line 299 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[1024]: File loader.c, Line 394 (load_modules): Loading module chan_zap.so failed! Whats wrong ? Thanks for help, Thomas. ___ 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] Sip too many open files?
Today my pbx stopped responding to my sip phones.. looking into the log, here what I got: Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new): Unable to allocate socket: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc): Unable to create RTP session: Too many open files Jun 23 15:50:05 WARNING[7176]: File chan_sip.c, Line 4655 (sip_send_mwi_to_peer): Unable to build sip pvt data for MWI Jun 23 15:51:07 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new): Unable to allocate socket: Too many open files Jun 23 15:51:07 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc): Unable to create RTP session: Too many open files Jun 23 15:51:07 WARNING[7176]: File chan_sip.c, Line 4655 (sip_send_mwi_to_peer): Unable to build sip pvt data for MWI Jun 23 15:51:07 WARNING[7176]: File channel.c, Line 293 (ast_channel_alloc): Alert pipe creation failed! Jun 23 15:51:07 WARNING[7176]: File chan_sip.c, Line 1152 (sip_new): Unable to allocate channel structure Jun 23 15:51:07 NOTICE[7176]: File chan_sip.c, Line 4414 (handle_request): Unable to create/find channel Jun 23 15:53:14 WARNING[7176]: File channel.c, Line 293 (ast_channel_alloc): Alert pipe creation failed! Jun 23 15:53:14 WARNING[7176]: File chan_sip.c, Line 1152 (sip_new): Unable to allocate channel structure Jun 23 15:53:14 NOTICE[7176]: File chan_sip.c, Line 4414 (handle_request): Unable to create/find channel Jun 23 15:53:34 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new): Unable to allocate socket: Too many open files Jun 23 15:53:34 WARNING[7176]: File chan_sip.c, Line 1308 (sip_alloc): Unable to create RTP session: Too many open files Jun 23 15:53:34 WARNING[7176]: File chan_sip.c, Line 4655 (sip_send_mwi_to_peer): Unable to build sip pvt data for MWI Jun 23 15:53:34 WARNING[7176]: File channel.c, Line 293 (ast_channel_alloc): Alert pipe creation failed! Jun 23 15:53:34 WARNING[7176]: File chan_sip.c, Line 1152 (sip_new): Unable to allocate channel structure Also wasn't possible to connect via a unix console... And so on... until I restarted the asterisk proc. What can cause that? I'm running CVS-06/22/03-16:32:23 , on a p4 2.4ghz and 512 mb ram, kern 2.4.21 Thanks a lot, Matteo -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -BEGIN GEEK CODE BLOCK- Version: 3.12 GS d? s:- a- C+++ UL P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e h! r++ y --END GEEK CODE BLOCK-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting up the E100P
THat's not it. in zapata.conf you *also* need to have signalling=pri_cpe or pri_net Martin On Mon, 23 Jun 2003, Michael Bielicki wrote: On Monday 23 June 2003 2:58 pm, Anton Yurchenko wrote: Hello, I have an E100P, and in the zaptel.conf I have: span=1,1,0,ccs,hdb4,crc4,yellow fxsks=1-10 delete the fxsks line and put: bchan=1-15,17-31 dchan=16 the light on the card is green( BTW what do all those states of the card that zttool reports YELLO, RED, BLUE ..., is there a doc for zttool?, or for the card?) in the asterisks` zapata.conf I have: [channels] context=default switchtype=euroisdn 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 callgroup=1 pickupgroup=1 immediate=no callerid=line1 238-20-31 channel = 1 callerid=line2 238-20-31 channel = 2 but Asterisk on startup reports that: WARNING[13326]: File chan_zap.c, Line 4173 (ss_thread): CallerID returned with rror on channel 'Zap/1-1' WARNING[14351]: File chan_zap.c, Line 4173 (ss_thread): CallerID returned with rror on channel 'Zap/2-1' and when I call the number that is supposed to terminate there I get busy signal. also in the zttool I see strange thing, in the bottom portion where there are RxA, RxB the 111 ( ones) sometimes change into 00, they go from up to down and if I enable all the channels then firs on the first column then on the second column, in the bottom half of the screen. while if I plug the same E1 into Cisco AS5300, whit this config( just exepts): isdn switch-type primary-net5 controller e1 ... clock source line primary pri-group timeslots 1-31 ... interface Serial0:15 isdn switch-type primary-net5 isdn incoming-voice modem 64 Framing is CRC4, Line Code is HDB3, Clock Source is Line ... on Cisco the swithtype is primary-net5 ( my guess its euroisdn ? ) anybody could guess what is the problem? The admin that runs the Cisco says that signalyng should be PRI , and there is an option for pri signaling in zapata.conf, but the zaptel,conf doesnt have it and so I`d get up with a mismatch, and zasterisk would not start. ___ 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] databases for billing
On Mon, 2003-06-23 at 03:03, carlos del mayor wrote: TWO THINGS,GARY! 1-Sorry for the html, now it's off 2-The mail you're talking about has arrived two minutes ago.I KNOW read, thank you. I only wanted to know if somebody was working with this, in order to simplify a litle my work (documentation and all that it's what i was looking for). Gary, I don't think your reaction is ok, people who are starting with new features (like me) only demand a litle of help and patience, only that. I know that we are a lot, and there are less 'guru' pepople than newbie one, but a mailing list is used to help. You can 'not help', I don't mind, but please don't send messages that can make newbie people feel this mailing list is not for them. The problem is demonstrated by your choice of words above. The rift between newbies and those with a little more knowledge is that newbies demand their questions be answered. The knowledge that is being demanded was not inexpensive for those who know it. We have either spent many hours learning, or for a few here many hours creating it. You do not see the same problems between gurus because we understand the value of the information that we are asking for, and how to communicate better with the person with the knowledge. Please understand that many times you will run into answers that are just one word, or one file name that is intended to point you to how to figure out your problem on your own. The end result is often that you gain your knowledge in a similar way that the person offering it had to. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk CPU power requirements
Many thanks, Martin .. worked fine with dtmfmode=info Dave - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, June 23, 2003 4:32 PM Subject: Re: [Asterisk-Users] Asterisk CPU power requirements You need to find out which way your SIP gateway wants to receive the DTMFs. There are three ways to do that. Read sip.conf.sample. Martin On Mon, 23 Jun 2003, Dave Alan Caruana wrote: hi there, I have an installed working Asterisk server, which I am using to connect to a SIP service abroad. Although I can hear the IVR from the ITSP, I cannot seem to send them digits from my phone. I have also noticed that the CPU usage on my machine is up to 100% constantly and 99.9% of that is going to Asterisk, even when asterisk is just idle and doing nothing at all .. The machine is a Celeron 800 with 256Mb of RAM, and there is a Digium single span E1 card going into it. Is something wrong? or do I just need more CPU power? cheers 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 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Please Help: Trying to build Asterisk - bazillionsof errors
On Sun, 22 Jun 2003, Steve wrote: Make sure you have the following installed: bison, cvs, gcc, kernel-source, libtermcap-devel, ncurses-devel, newt-devel, openssl096b, openssl-devel, readline and readline-devel. readline and readline-devel have not been needed since November of last year. James ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Active ISDN PCMCIA card
Thanks for the replies. It seems that AVM B1 is the only active PCMCIA card that can be used with Asterisk. The kernel supports this card, so I guess that the driver can be built on non-x86 systems. Regards, Michael. Olaf Menzel wrote: On Friday 20 June 2003 13:28, Michael Manousos wrote: Are there any suggestions for active ISDN CAPI PCMCIA cards that are known to work with Asterisk? You can try AVM B1 PCMCIA. This card is fully I4L compliant but AVM has developed a LINUX capi 2.0 stack. http://www.avm.de/en/products/hardware/active/B1_PCMCIA/index.html The Linux Capi driver you find here: ftp://ftp.avm.de/cardware/b1_pcm/linux/ Be aware that the Capi4Linux driver is distributed only as binary and especially prepared for Suse distributions. WIth some adaptations it should work with other distributions as well. Otherwise you should use I4L for this card. BTW. The Capi4Linux driver works also for the AVM Fritz which is much cheaper than the B1 device and supports full CAPI functionality such as G3 Fax. regards Olaf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip too many open files?
On Monday 23 June 2003 09:32 am, Brancaleoni Matteo wrote: Today my pbx stopped responding to my sip phones.. looking into the log, here what I got: Jun 23 15:50:05 WARNING[7176]: File rtp.c, Line 586 (ast_rtp_new): Unable to allocate socket: Too many open files The open file limit is per user, so you should still be able to login as any user the asterisk process is not running as. Once you're at that point, run 'lsof' to see what files are open. It's possible that they're all sockets which aren't getting shutdown(2) properly. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Module app_perl
Would anyone be so kind as to explain why no voice is heard through the phone when calling? Thanks. On Mon, 2003-06-23 at 10:34, James Golovich wrote: No point in reinventing the wheel here. PersistentPerl (aka SpeedyCGI) can eliminate the startup cost for using perl with AGIs. It works great, and even allows the processes to reuse database connections James On Mon, 23 Jun 2003, Anthony Minessale wrote: That is probably possible and not too difficult. I learned what AGI was about 30 minutes after I was finished with the last revision of app_perl where I added support to launch a perl function in a thread (BTW I am suspicious that you may ironically need perl with no threads compiled for it to work right in asterisk despite the fact that you gain thread functionality via asterisk) I have not really carefully looked at AGI yet but from what I remember It communicates with the ext process via STDIN and looks at the ENV for information. so what I think you would need would be a fake ENV and a special variable to contain the same info that would have been sent to STDIN created uniquely for each execution. This of course would be limiting the AGI to perl code so another method would be to make a function via the app_perl or a dedicated C module to run all the agi app at startup and leave them open speaking back and forth over IO stream. I do notice I started stepping on the toes of AGI because I never heard of it while I was coding my module so I think some of the things that AGI does can also be accomplished on app_perl The 3 things I was dreaming of when I was working on it were: 1) If the module has the power to create extensions then you can use it to fetch that data from a database on startup or in mid run. 2) If the module can run threads It could implement an external listener of some sort and communicate with a partner thread over shared memory and with the world over sockets, tcpip etc (web server) I had a demo working where you could go to the asterisk on a web browser an see a readout like number is 0 then if you dial into a certain ext with a phone the number increments and when you reload the web page it said number is 1 and so on. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Jordan In a world without windows, who needs gates? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gastman and New Extension
I finally got Gastman to compile but I get a bunch of failed assertions when I run it and attempt to make a new extension. I have latest CVS on Mandrake 9.1. Last error is: (gastman:22534): Gdk-CRITICAL **: file ../../gdk/gdkdraw.c: line 311 (gdk_drawable_unref): assertion `GDK_IS_DRAWABLE (drawable)' failed Other errors are like this one. I apologize if this has been addressed previously but a quick search in the list turned up nothing. Any quick ideas? Jim Friedeck ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk hogging CPU resources
What appears to be hogging CPU? What interfaces are you running? Mark On Fri, 20 Jun 2003, Derek Beaumont wrote: Here's the problem: I start asterisk, and it takes up around 3-4% of my CPU resources. However, this number continues to climb over the hours until it is close to 100%. Usually it takes around a day to climb up to approximately 95 or 96% Has anybody experienced the following problem before? ___ 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] Manager interface, again
If in your voicemail.conf you have * configured to the send message in an email you will NOT get a stutter dialtone or any MWI light you may have. I've just removed my email address from voicemail.conf.. much better like that... I can't see how that would make any difference. Can you find me on IRC so I can ssh in and try to see what's going on? Thanks! HTHSITF What's that mean? Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Module app_perl
replying to 2 other threads with your problem is not the way to get people to answer your question. If you search the archive you will see that voice modems are not really supported. This is why you don't hear audio. Now quit being impatient and _DEMANDING_ support. On Mon, 2003-06-23 at 13:07, Jordan Peterson wrote: Would anyone be so kind as to explain why no voice is heard through the phone when calling? Thanks. On Mon, 2003-06-23 at 10:34, James Golovich wrote: No point in reinventing the wheel here. PersistentPerl (aka SpeedyCGI) can eliminate the startup cost for using perl with AGIs. It works great, and even allows the processes to reuse database connections James On Mon, 23 Jun 2003, Anthony Minessale wrote: That is probably possible and not too difficult. I learned what AGI was about 30 minutes after I was finished with the last revision of app_perl where I added support to launch a perl function in a thread (BTW I am suspicious that you may ironically need perl with no threads compiled for it to work right in asterisk despite the fact that you gain thread functionality via asterisk) I have not really carefully looked at AGI yet but from what I remember It communicates with the ext process via STDIN and looks at the ENV for information. so what I think you would need would be a fake ENV and a special variable to contain the same info that would have been sent to STDIN created uniquely for each execution. This of course would be limiting the AGI to perl code so another method would be to make a function via the app_perl or a dedicated C module to run all the agi app at startup and leave them open speaking back and forth over IO stream. I do notice I started stepping on the toes of AGI because I never heard of it while I was coding my module so I think some of the things that AGI does can also be accomplished on app_perl The 3 things I was dreaming of when I was working on it were: 1) If the module has the power to create extensions then you can use it to fetch that data from a database on startup or in mid run. 2) If the module can run threads It could implement an external listener of some sort and communicate with a partner thread over shared memory and with the world over sockets, tcpip etc (web server) I had a demo working where you could go to the asterisk on a web browser an see a readout like number is 0 then if you dial into a certain ext with a phone the number increments and when you reload the web page it said number is 1 and so on. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ringing tones oh323
When i make a call using oh323 channels, how i can send a ringing sounds to indicate to the users that the call is in progress thanks
[Asterisk-Users] unsubscribe
- Original Message - From: Jorge Cisneros To: [EMAIL PROTECTED] Sent: Monday, June 23, 2003 3:57 PM Subject: [Asterisk-Users] Ringing tones oh323 When i make a call using oh323 channels, how i can send a ringing sounds to indicate to the users that the call is in progress thanks
Re: [Asterisk-Users] New Module app_perl
Jerk On Mon, 2003-06-23 at 13:02, Steven Critchfield wrote: replying to 2 other threads with your problem is not the way to get people to answer your question. If you search the archive you will see that voice modems are not really supported. This is why you don't hear audio. Now quit being impatient and _DEMANDING_ support. On Mon, 2003-06-23 at 13:07, Jordan Peterson wrote: Would anyone be so kind as to explain why no voice is heard through the phone when calling? Thanks. On Mon, 2003-06-23 at 10:34, James Golovich wrote: No point in reinventing the wheel here. PersistentPerl (aka SpeedyCGI) can eliminate the startup cost for using perl with AGIs. It works great, and even allows the processes to reuse database connections James On Mon, 23 Jun 2003, Anthony Minessale wrote: That is probably possible and not too difficult. I learned what AGI was about 30 minutes after I was finished with the last revision of app_perl where I added support to launch a perl function in a thread (BTW I am suspicious that you may ironically need perl with no threads compiled for it to work right in asterisk despite the fact that you gain thread functionality via asterisk) I have not really carefully looked at AGI yet but from what I remember It communicates with the ext process via STDIN and looks at the ENV for information. so what I think you would need would be a fake ENV and a special variable to contain the same info that would have been sent to STDIN created uniquely for each execution. This of course would be limiting the AGI to perl code so another method would be to make a function via the app_perl or a dedicated C module to run all the agi app at startup and leave them open speaking back and forth over IO stream. I do notice I started stepping on the toes of AGI because I never heard of it while I was coding my module so I think some of the things that AGI does can also be accomplished on app_perl The 3 things I was dreaming of when I was working on it were: 1) If the module has the power to create extensions then you can use it to fetch that data from a database on startup or in mid run. 2) If the module can run threads It could implement an external listener of some sort and communicate with a partner thread over shared memory and with the world over sockets, tcpip etc (web server) I had a demo working where you could go to the asterisk on a web browser an see a readout like number is 0 then if you dial into a certain ext with a phone the number increments and when you reload the web page it said number is 1 and so on. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Jordan In a world without windows, who needs gates? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with native bridge function.
Hi all, I have problems with native bridging with this configuration; CPE(Mediatrix SIP-G.729)-Asterisk-Cisco AS5300 (SIP-G.729) Problem is, remote side get very bad sound while local end is getting very clear quality. If I set below configuration and make asterisk to encode CPE(Mediatrix SIP-ULAW)-Asterisk-Cisco AS5300 (SIP-G.729) Sound is very good in both directions. My configuration is; Pentium IV 1.6 Ghz Suse 7.1 512 MB Ram Asterisk 0.3 and one g.729 license from digium. Yours Sincerely
Re: [Asterisk-Users] New Module app_perl
On Monday 23 June 2003 03:24 pm, Jordan Peterson wrote: Jerk And one who is contributing to the development of Asterisk. If you aren't the patient type and would like immediate answers to your questions, I strongly advise calling Digium and buying a support contract. The support techs are very patient and will be more than happy to help you. In this forum, however, you're getting free support, so be prepared to take whatever anybody throws at you. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 911/Emergency calls + Caller ID
Problem: 911 calls placed through Asterisk are associated with the physical location of where the CO trunks terminate. This is not really a problem when all extensions are located in the same building, but when Asterisk is used in a campus-like or otherwise networked environment, it can get messy. A common solution is to install a few analog lines at each location, for emergency calls only. But by making clever use of Caller ID (and adding a 'location' field to extensions.conf), it should be possible to properly identify the location of the caller: exten = 1001,1,John Doe,1223 Bell Ave. Room 51 For this to work, you would have to be able to apply rules to the 911 context in a dial plan, to replace the *name* portion with the *location* portion. A similar rule could be defined to drop other calls if 911 is dialed and all lines are busy (e.g. drop the lobby phone but not the front desk, or drop local vs. long distance, caller ID calls vs. non-identified calls, etc.). Getting lengthy, better stop. Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Module app_perl
Remove the space behind .com, like so http://asterisk.650dialup.com/ Cheers, Dylan. Uriel Carrasquilla wrote: For some reason the page cannot be found. http://asterisk.650dialup.com what does it do? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ask Bjørn Hansen Sent: Monday, June 23, 2003 5:12 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Module app_perl On Tuesday, Jun 17, 2003, at 20:43 America/Los_Angeles, Anthony Minessale wrote: Here is a copy of the first release (comments appreciated) http://asterisk.650dialup.com Although I haven't had time to play with it: very neat! - ask -- http://www.askbjoernhansen.com/ ___ 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] dynamic queue channels
Hi, Im trying to build a call center application that allows attendants to come in the morning and dial a certain extension to make their extension available. I wouldnt like to use the AgentLogin app because their line would need to stay off-hook (is this correct?) Is there any SET channel status command that would allow me to do something like this? PauloHM
Re: [Asterisk-Users] 911/Emergency calls + Caller ID
I'm not sure I can parse your examples correctly. I'm not being snide, but do you use Asterisk on a regular basis? Do you understand how applications work, and how call handoff is done between Asterisk servers? Your example doesn't seem to make sense, no matter how I think about it. Of course, the problem with 911 is the problem of location of the originating handset. That much has been clear for years. Getting that information to the 911 call center is the problem; it's pretty much worthless info even if you have it inside the PBX - you could just as easily have an external database that maps extensions to locations - why bother with the PBX if there is no in-band signalling to the PSAP? This makes me think a bit about some other 911 ideas I had a while back, using lat/lon/altitude. Can ADSI tones be transmitted through any phone call on the PSTN? It might be interesting for PBX systems to pass across the lat/lon/altitude of callers via ADSI in-band. This will never work, of course, since nobody would trust the transmitters. The 911 question almost instantly spins into a political issue, and not a technical issue, since there are a number of clever ways to solve the problem but not a number of clever ways to bang solutions into people's heads. Bumping calls to clear a path for 911 is possible within Asterisk already - see the SoftHangup application. JT Problem: 911 calls placed through Asterisk are associated with the physical location of where the CO trunks terminate. This is not really a problem when all extensions are located in the same building, but when Asterisk is used in a campus-like or otherwise networked environment, it can get messy. A common solution is to install a few analog lines at each location, for emergency calls only. But by making clever use of Caller ID (and adding a 'location' field to extensions.conf), it should be possible to properly identify the location of the caller: exten = 1001,1,John Doe,1223 Bell Ave. Room 51 For this to work, you would have to be able to apply rules to the 911 context in a dial plan, to replace the *name* portion with the *location* portion. A similar rule could be defined to drop other calls if 911 is dialed and all lines are busy (e.g. drop the lobby phone but not the front desk, or drop local vs. long distance, caller ID calls vs. non-identified calls, etc.). Getting lengthy, better stop. Dylan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911/Emergency calls + Caller ID
Now that I reed it back, I can barely make sense of it myself! Anyway, I was just thinking out loud, the example wasn't meant to be parsed. Asterisk would need some lower level changes to parse the extra field holding the location information, and to apply the routing rules to substitute the Caller ID name for the location. I was hoping this would be thought provoking for somebody smarter than me :) Bumping calls to clear a path for 911 is possible within Asterisk already - see the SoftHangup application. That sounds good, but what can trigger the SoftHangup app to drop other calls automatically when 911 is dialed? Thanks, Dylan. John Todd wrote: I'm not sure I can parse your examples correctly. I'm not being snide, but do you use Asterisk on a regular basis? Do you understand how applications work, and how call handoff is done between Asterisk servers? Your example doesn't seem to make sense, no matter how I think about it. Of course, the problem with 911 is the problem of location of the originating handset. That much has been clear for years. Getting that information to the 911 call center is the problem; it's pretty much worthless info even if you have it inside the PBX - you could just as easily have an external database that maps extensions to locations - why bother with the PBX if there is no in-band signalling to the PSAP? This makes me think a bit about some other 911 ideas I had a while back, using lat/lon/altitude. Can ADSI tones be transmitted through any phone call on the PSTN? It might be interesting for PBX systems to pass across the lat/lon/altitude of callers via ADSI in-band. This will never work, of course, since nobody would trust the transmitters. The 911 question almost instantly spins into a political issue, and not a technical issue, since there are a number of clever ways to solve the problem but not a number of clever ways to bang solutions into people's heads. Bumping calls to clear a path for 911 is possible within Asterisk already - see the SoftHangup application. JT Problem: 911 calls placed through Asterisk are associated with the physical location of where the CO trunks terminate. This is not really a problem when all extensions are located in the same building, but when Asterisk is used in a campus-like or otherwise networked environment, it can get messy. A common solution is to install a few analog lines at each location, for emergency calls only. But by making clever use of Caller ID (and adding a 'location' field to extensions.conf), it should be possible to properly identify the location of the caller: exten = 1001,1,John Doe,1223 Bell Ave. Room 51 For this to work, you would have to be able to apply rules to the 911 context in a dial plan, to replace the *name* portion with the *location* portion. A similar rule could be defined to drop other calls if 911 is dialed and all lines are busy (e.g. drop the lobby phone but not the front desk, or drop local vs. long distance, caller ID calls vs. non-identified calls, etc.). Getting lengthy, better stop. Dylan. ___ 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] 911/Emergency calls + Caller ID
Dylan VanHerpen wrote: Now that I reed it back, I can barely make sense of it myself! Anyway, I was just thinking out loud, the example wasn't meant to be parsed. Asterisk would need some lower level changes to parse the extra field holding the location information, and to apply the routing rules to substitute the Caller ID name for the location. I was hoping this would be thought provoking for somebody smarter than me :) Bumping calls to clear a path for 911 is possible within Asterisk already - see the SoftHangup application. That sounds good, but what can trigger the SoftHangup app to drop other calls automatically when 911 is dialed? Thanks, Dylan. John Todd wrote: I'm not sure I can parse your examples correctly. I'm not being snide, but do you use Asterisk on a regular basis? Do you understand how applications work, and how call handoff is done between Asterisk servers? Your example doesn't seem to make sense, no matter how I think about it. Of course, the problem with 911 is the problem of location of the originating handset. That much has been clear for years. Getting that information to the 911 call center is the problem; it's pretty much worthless info even if you have it inside the PBX - you could just as easily have an external database that maps extensions to locations - why bother with the PBX if there is no in-band signalling to the PSAP? This makes me think a bit about some other 911 ideas I had a while back, using lat/lon/altitude. Can ADSI tones be transmitted through any phone call on the PSTN? It might be interesting for PBX systems to pass across the lat/lon/altitude of callers via ADSI in-band. This will never work, of course, since nobody would trust the transmitters. The 911 question almost instantly spins into a political issue, and not a technical issue, since there are a number of clever ways to solve the problem but not a number of clever ways to bang solutions into people's heads. Bumping calls to clear a path for 911 is possible within Asterisk already - see the SoftHangup application. JT Problem: 911 calls placed through Asterisk are associated with the physical location of where the CO trunks terminate. This is not really a problem when all extensions are located in the same building, but when Asterisk is used in a campus-like or otherwise networked environment, it can get messy. A common solution is to install a few analog lines at each location, for emergency calls only. But by making clever use of Caller ID (and adding a 'location' field to extensions.conf), it should be possible to properly identify the location of the caller: exten = 1001,1,John Doe,1223 Bell Ave. Room 51 For this to work, you would have to be able to apply rules to the 911 context in a dial plan, to replace the *name* portion with the *location* portion. A similar rule could be defined to drop other calls if 911 is dialed and all lines are busy (e.g. drop the lobby phone but not the front desk, or drop local vs. long distance, caller ID calls vs. non-identified calls, etc.). Getting lengthy, better stop. Dylan. ___ 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] 911/Emergency calls + Caller ID
And now that I *read* it back again, you can tell that English is not my native language either Dylan VanHerpen wrote: Now that I reed it back, I can barely make sense of it myself! Anyway, I was just thinking out loud, the example wasn't meant to be parsed. Asterisk would need some lower level changes to parse the extra field holding the location information, and to apply the routing rules to substitute the Caller ID name for the location. I was hoping this would be thought provoking for somebody smarter than me :) Bumping calls to clear a path for 911 is possible within Asterisk already - see the SoftHangup application. That sounds good, but what can trigger the SoftHangup app to drop other calls automatically when 911 is dialed? Thanks, Dylan. John Todd wrote: I'm not sure I can parse your examples correctly. I'm not being snide, but do you use Asterisk on a regular basis? Do you understand how applications work, and how call handoff is done between Asterisk servers? Your example doesn't seem to make sense, no matter how I think about it. Of course, the problem with 911 is the problem of location of the originating handset. That much has been clear for years. Getting that information to the 911 call center is the problem; it's pretty much worthless info even if you have it inside the PBX - you could just as easily have an external database that maps extensions to locations - why bother with the PBX if there is no in-band signalling to the PSAP? This makes me think a bit about some other 911 ideas I had a while back, using lat/lon/altitude. Can ADSI tones be transmitted through any phone call on the PSTN? It might be interesting for PBX systems to pass across the lat/lon/altitude of callers via ADSI in-band. This will never work, of course, since nobody would trust the transmitters. The 911 question almost instantly spins into a political issue, and not a technical issue, since there are a number of clever ways to solve the problem but not a number of clever ways to bang solutions into people's heads. Bumping calls to clear a path for 911 is possible within Asterisk already - see the SoftHangup application. JT Problem: 911 calls placed through Asterisk are associated with the physical location of where the CO trunks terminate. This is not really a problem when all extensions are located in the same building, but when Asterisk is used in a campus-like or otherwise networked environment, it can get messy. A common solution is to install a few analog lines at each location, for emergency calls only. But by making clever use of Caller ID (and adding a 'location' field to extensions.conf), it should be possible to properly identify the location of the caller: exten = 1001,1,John Doe,1223 Bell Ave. Room 51 For this to work, you would have to be able to apply rules to the 911 context in a dial plan, to replace the *name* portion with the *location* portion. A similar rule could be defined to drop other calls if 911 is dialed and all lines are busy (e.g. drop the lobby phone but not the front desk, or drop local vs. long distance, caller ID calls vs. non-identified calls, etc.). Getting lengthy, better stop. Dylan. ___ 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] New Module app_perl
Great work! Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dylan VanHerpen Sent: Monday, June 23, 2003 7:26 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Module app_perl Remove the space behind .com, like so http://asterisk.650dialup.com/ Cheers, Dylan. Uriel Carrasquilla wrote: For some reason the page cannot be found. http://asterisk.650dialup.com what does it do? Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ask Bjørn Hansen Sent: Monday, June 23, 2003 5:12 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Module app_perl On Tuesday, Jun 17, 2003, at 20:43 America/Los_Angeles, Anthony Minessale wrote: Here is a copy of the first release (comments appreciated) http://asterisk.650dialup.com Although I haven't had time to play with it: very neat! - ask -- http://www.askbjoernhansen.com/ ___ 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] 911/Emergency calls + Caller ID
Problem: 911 calls placed through Asterisk are associated with the physical location of where the CO trunks terminate. This is not really a problem when all extensions are located in the same building, but when Asterisk is used in a campus-like or otherwise networked environment, it can get messy. A common solution is to install a few analog lines at each location, for emergency calls only. But by making clever use of Caller ID (and adding a 'location' field to extensions.conf), it should be possible to properly identify the location of the caller: exten = 1001,1,John Doe,1223 Bell Ave. Room 51 For this to work, you would have to be able to apply rules to the 911 context in a dial plan, to replace the *name* portion with the *location* portion. A similar rule could be defined to drop other calls if 911 is dialed and all lines are busy (e.g. drop the lobby phone but not the front desk, or drop local vs. long distance, caller ID calls vs. non-identified calls, etc.). Getting lengthy, better stop. Dylan. This is all quite interesting to me, as I have been somewhat concerned about it, though have never quite bumped into it directly yet. It would be 'nice' to be able to forcibly hangup on some rule based channel if a certain dial 'priority' is set. Perhaps you could do something like this: exten = 911,1,SetVar(priority,911) exten = 911,2,Dial,Zap/g2:911 (Ignore the likely invalid syntax/parameters, but you should get the right idea) Then in another config file: [911] On,Busy,Drop,Zap/1 On,Busy,Drop,Zap/g2 On,Busy,Drop,any So, we might initially try hanging up on Zap/1, but for some reason, we can't release the channel, so we now try each line in Zap/g2 successively, if we still can't get a channel to become available, then try any other line, (heck, drop all of them and pickup the first available). You could also use this so that just before your boss's dialout, it sets the priority to 666, the first thing it does is try to disconnect the line your extension is using (because you only ever talk to your friend and spend all day chatting instead of working, but don't want your boss to realise you were on the phone again...) Yes, it is possible to use SoftHangup to do this, it can be done as an AGI, but I think the importance of this is such that the level of peer review and correctness is rather high! Imagine you get it wrong and all it does is hang up on the caller when they dial 911. Also, it isn't very easy to 'test' either, as the staff at the 911 call centre won't appreciate your testing, and at least in Australia, it is some sort of criminal?/illegal offence to call emergency for non-emergency situations. PS, also keep in mind that different countries use different codes for emergency. Personally, when setting up these codes, I have tried to accomodate for all the ones I know of: 911 - North America 000 - Australia 112 - Emergency from Mobile Phones in Australia I'm not sure what the number is in other countries, but perhaps we should allow this to be somewhat flexible enough that it can be used anywhere. Just some additional lengthy comments to add to the list :) Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 911/Emergency calls + Caller ID
Also, it isn't very easy to 'test' either, as the staff at the 911 call centre won't appreciate your testing, and at least in Australia, it is some sort of criminal?/illegal offence to call emergency for non-emergency situations. I had much the same thoughts. Currently my 911 code is just commented out for that very reason - I don't want to get in trouble for accidentally making 911 calls to test it. Should I rely on that code untested for when it is really needed most ? What are other people doing ? I have a set of extensions I call line seize that are supposed to act like the line buttons on a conventional business phone to pickup a specific line and get a dial tone (I was going to add them to adsi to make the illusion even more complete), maybe I will modify those to include a softhangup when the line is busy if the user hits * or something. In a real emergency though you would want this as simple as possible, but foolproof if you code it wrong. PS, also keep in mind that different countries use different codes for emergency. Personally, when setting up these codes, I have tried to accomodate for all the ones I know of: 911 - North America 000 - Australia 112 - Emergency from Mobile Phones in Australia I'm not sure what the number is in other countries, but perhaps we should allow this to be somewhat flexible enough that it can be used anywhere. Just some additional lengthy comments to add to the list :) Regards, Adam ___ 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] 911/Emergency calls + Caller ID
Jon Pounder wrote: I had much the same thoughts. Currently my 911 code is just commented out for that very reason - I don't want to get in trouble for accidentally making 911 calls to test it. Should I rely on that code untested for when it is really needed most ? What are other people doing ? Cisco have implemented a solution for this, does anyone know how they do it in Call Manager? -- Regards, David Hooton Senior Partner Platform Hosting www.platformhosting.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911/Emergency calls + Caller ID
Also, it isn't very easy to 'test' either, as the staff at the 911 call centre won't appreciate your testing, and at least in Australia, it is some sort of criminal?/illegal offence to call emergency for non-emergency situations. Well, for testing purposes 911 could be replaced with any other number. You can also setup an alias for '11', so that regardless if people dial 911 (instead of 9,911), they'll get thru. Dylan. Adam Goryachev wrote: Problem: 911 calls placed through Asterisk are associated with the physical location of where the CO trunks terminate. This is not really a problem when all extensions are located in the same building, but when Asterisk is used in a campus-like or otherwise networked environment, it can get messy. A common solution is to install a few analog lines at each location, for emergency calls only. But by making clever use of Caller ID (and adding a 'location' field to extensions.conf), it should be possible to properly identify the location of the caller: exten = 1001,1,John Doe,1223 Bell Ave. Room 51 For this to work, you would have to be able to apply rules to the 911 context in a dial plan, to replace the *name* portion with the *location* portion. A similar rule could be defined to drop other calls if 911 is dialed and all lines are busy (e.g. drop the lobby phone but not the front desk, or drop local vs. long distance, caller ID calls vs. non-identified calls, etc.). Getting lengthy, better stop. Dylan. This is all quite interesting to me, as I have been somewhat concerned about it, though have never quite bumped into it directly yet. It would be 'nice' to be able to forcibly hangup on some rule based channel if a certain dial 'priority' is set. Perhaps you could do something like this: exten = 911,1,SetVar(priority,911) exten = 911,2,Dial,Zap/g2:911 (Ignore the likely invalid syntax/parameters, but you should get the right idea) Then in another config file: [911] On,Busy,Drop,Zap/1 On,Busy,Drop,Zap/g2 On,Busy,Drop,any So, we might initially try hanging up on Zap/1, but for some reason, we can't release the channel, so we now try each line in Zap/g2 successively, if we still can't get a channel to become available, then try any other line, (heck, drop all of them and pickup the first available). You could also use this so that just before your boss's dialout, it sets the priority to 666, the first thing it does is try to disconnect the line your extension is using (because you only ever talk to your friend and spend all day chatting instead of working, but don't want your boss to realise you were on the phone again...) Yes, it is possible to use SoftHangup to do this, it can be done as an AGI, but I think the importance of this is such that the level of peer review and correctness is rather high! Imagine you get it wrong and all it does is hang up on the caller when they dial 911. Also, it isn't very easy to 'test' either, as the staff at the 911 call centre won't appreciate your testing, and at least in Australia, it is some sort of criminal?/illegal offence to call emergency for non-emergency situations. PS, also keep in mind that different countries use different codes for emergency. Personally, when setting up these codes, I have tried to accomodate for all the ones I know of: 911 - North America 000 - Australia 112 - Emergency from Mobile Phones in Australia I'm not sure what the number is in other countries, but perhaps we should allow this to be somewhat flexible enough that it can be used anywhere. Just some additional lengthy comments to add to the list :) Regards, Adam ___ 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] 911/Emergency calls + Caller ID
Also, it isn't very easy to 'test' either, as the staff at the 911 call centre won't appreciate your testing, and at least in Australia, it is some sort of criminal?/illegal offence to call emergency for non-emergency situations. I had much the same thoughts. Currently my 911 code is just commented out for that very reason - I don't want to get in trouble for accidentally making 911 calls to test it. Should I rely on that code untested for when it is really needed most ? What are other people doing ? In my experience, most 911 operators will say thank you, hang up, and go about their business if you tell them as soon as they answer the phone that This is a telephone system test call to ensure 911 operation. Most of all, don't hang up on them when they answer or you'll have a patrol car sitting at your place soon after. As long as you don't call them every 10 minutes, it shouldn't be a problem. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 911/Emergency calls + Caller ID
Bumping calls to clear a path for 911 is possible within Asterisk already - see the SoftHangup application. That sounds good, but what can trigger the SoftHangup app to drop other calls automatically when 911 is dialed? A short AGI script, perhaps? It probably would not even require a short AGI. Define a group of Zap lines as your emergency lines. Increment a counter every time a line in that group is used for an outbound/inbound call, and decrement when the line is released (hung up.) If a 911 call is placed, and counter=(max lines in group) then run the SoftHangup and hangup the last three or four lines in the group before placing the 911 call. It is hopefully the case that your system sees 911 calls infrequently enough that a few dropped calls will not be overly burdensome. A sub-counter needs to be kept in order to prevent an existing 911 call from being SoftHangup'ed. It is the case that 911 calls come in clusters from office environments, where two or three people may call about the same issue at the same time, and it would be bad form to hang up 911 caller #1 in order to clear the line for 911 caller #2. You simply have to judge how many lines are appropriate For simplicity's sake, you may just decide that you should hang up Zap/1-21, Zap/1-22, Zap/1-23 anytime you see a 911 call being placed. You risk hanging up on your other 911 callers... but everything is always a tradeoff. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
