Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
On Wednesday 19 January 2005 23:15, Eric Bishop wrote: Well guys this is truly bizarre. I managed to get a DL360 G3 to show interrupts with FC2 but not FC3. Exact same config and setup proceedure. Ofcourse neither FC2 or FC3 show interrupts with the DL360 G4. I think TE410P is just a flakey card. Anyone got a DL360 G3 going with a TE410P and FC3? I did manage to get a TE110P running on the DL380 G4. Still can't get the TE410P working in the G4 though. Supports your theory. Sadly we're now being forced to look elsewhere for PRI cards. -- Regards, Tais M. Hansen ComX Networks A/S Tel: +45-70257474 Fax: +45-70257374 pgpL97AQuIMRT.pgp Description: PGP signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Authentication Problem
Hello everbody, I am having problems is Database version and Real time version of Asterisk. Users are connecting with no problem, they gets authenticate and its working fine, but after 2-3 minutes, registration with the same user comes and it gets failed to authenticate. dial tone gone, users unable to call, but this behaviour not remains for the all users for all the time. most of the time they are able to call, its totally wiered to me. any ideas ? -Neo p.s i m using different kinds of clients xlite xpro cisco ata dta ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.3 startup
Hi All, I've managed to compile make and make install asterisk on Mandrake 9.2. However on startup I get the following message: [cdr_tds.so]Jan 20 11:13:54 WARNING[20999]: loader.c:258 ast_load_resource: libtds.so.3: cannot open shared object file: No such file or directoryJan 20 11:13:54 WARNING[20999]: loader.c:440 load_modules: Loading module cdr_tds.so failed! I have freetds installed from RPM, and the lib is here /usr/local/lib/libtds.so.3.0.0 Where does asterisk look for the lib ? Maybe I can do a symlink ? Any help appreciated. Thanks and Regards Nic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Asterisk 1.0.3 startup
Sorry all, Did that and its going good now. Rgds Nic From: Nic le Roux [mailto:[EMAIL PROTECTED] Sent: 20 January 2005 11:22 AMTo: 'asterisk-users@lists.digium.com'Subject: Asterisk 1.0.3 startup Hi All, I've managed to compile make and make install asterisk on Mandrake 9.2. However on startup I get the following message: [cdr_tds.so]Jan 20 11:13:54 WARNING[20999]: loader.c:258 ast_load_resource: libtds.so.3: cannot open shared object file: No such file or directoryJan 20 11:13:54 WARNING[20999]: loader.c:440 load_modules: Loading module cdr_tds.so failed! I have freetds installed from RPM, and the lib is here /usr/local/lib/libtds.so.3.0.0 Where does asterisk look for the lib ? Maybe I can do a symlink ? Any help appreciated. Thanks and Regards Nic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] API Call Bridge?
Hi All, Does anyone know of a way to dial two different outbound numbers and bridge them together using the Asterisk API? Cheers, Taff. ALL-NEW Yahoo! Messenger - all new features - even more fun! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls
I would also suggest that while it is possible to do something, it is not always wise :) See the significant volumes of reports in the archives regarding multiple zaptel cards in one system. I must be lucky: I have 2 X100P and a TDM400 with zero IRQ or other issues. And double NAT for the voIP part. :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls
On Thu, 20 Jan 2005, Wilson Pickett wrote: I would also suggest that while it is possible to do something, it is not always wise :) See the significant volumes of reports in the archives regarding multiple zaptel cards in one system. I must be lucky: I have 2 X100P and a TDM400 with zero IRQ or other issues. And double NAT for the voIP part. :) Yeah - here's my 3 card system: A TE410P running three PRIs, and two TDM400Ps running total of 6 active FXO ports: # cat /proc/interrupts CPU0 0: 62571778IO-APIC-edge timer 1: 83IO-APIC-edge i8042 9: 0 IO-APIC-level acpi 14: 167849IO-APIC-edge ide0 15: 13IO-APIC-edge ide1 17:7537473 IO-APIC-level eth0 24: 62545476 IO-APIC-level wctdm 25: 62546044 IO-APIC-level wctdm 26: 62537638 IO-APIC-level t4xxp NMI: 0 LOC: 62572633 ERR: 0 Its an HP ML110. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to read ISDN messages - URGENT!!!!
Hi, Were using Asterisk with Digium TE110P card for the PSTN E1 interface. Our PRI is enabled with detecting the connected-party-number feature. When an OUTBOUND call is made to a phone, the PRI will send back an ISDN messages containing the connected-number and we can use that information to validate the extension user is calling the party that he/she is authorized to. This is to avoid the user letting know the receiver about the call and getting the receiver to divert the phone to some other number. How can we read this ISDN messages from Asterisk? Your help would be VERY much appreciated. Lilantha ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] API Call Bridge?
On Thu, 20 Jan 2005, taf taffey wrote: Does anyone know of a way to dial two different outbound numbers and bridge them together using the Asterisk API? Which api do you mean? There are at least two ways: - Using a call file in the spool directory - Using the originate command in the mangager api Both work the same way. This information is on the wiki... Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hardware details
Hello, I want to build a PBX with the following specs : 1. we have two trunk lines 2. we need upto 8 extensions - all analog phones maybe one voip phone What is hardware that I need and where to find it ? Thanks Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] regexten for realtime sip ?
Hi, sip.conf has a paramter regexten using which we can assign an extension to a registered SIP client and can use the same number to call that client. Is there any such parameter for realtime sip table sip_buddies. Why was this missed out in this table ? Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] (no subject)
Hello, Asterisk provides its own Asterisk gatekeeper is there other wise it supprots gnugk please tell me Thank u Sailatha[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Poor sound quality on ISDN BRI calls
I've been struggling with connection Asterisk to ISDN BRI lines for a while. I have it working with the latest bristuff and compatible Asterisk version: Asterisk 1.0.3-BRIstuffed-0.2.0-RC3a I am using a cheap Centronics ISDN card and the zaphfc drivers. It works but users complain that the sound quality is not good. They have Xlite phones on their desktops. Xlite to Xlite through Asterisk is fine. Xlite to PSTN through ISDN is not good. Anyone got any experience with this kind of setup and improving sound quality? I will add anything new info to the Wiki. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Advanced Agents - Need a nice web interface
Can anybody help me find this patch? So nobody knows of a pre-built web-interface that can accomplish these goals? Ohh well, time to work with a developer to custom build one. Anybody else interested in these features? Should I post the source/code once I have it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nathan Bowyer Sent: Wednesday, January 19, 2005 4:02 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Advanced Agents - Need a nice web interface On Wed, 19 Jan 2005 12:56:59 -0500, Paul Rodan [EMAIL PROTECTED] wrote: [snip] 3. Create historical report to pull agent activity. Should display login/logout activity. Be able to pull information by rep and timeframe. This could probably be done with the CDRs and queue_log. 4. Create hold calls/bypass statuses for agent login. This status should allow the rep to pause all incoming calls to their login for reasons such as: 1-Break, 2-Lunch, 3-Meeting, 4-Project, 5-Other. This status should not log the agent out of the phone, but only temporarily take them out of the queue to receive the next available call until they end the hold/bypass status and make themselves available for incoming calls. There was a patch in the bug tracker (bugs.digium.com) a week or so ago about pausing agents. It would temporarily stop calls coming to their station, but not log them out, as I recall. I'm thinking no, but I figured I'd ask anyways before telling my bosses they're out of their minds. Even if there's an existing interface out there that can provide 1 or 2 of these things, it'd be a nice start. Most of it I'd have to work with a developer to get created, and I'm thinking option 4 is impossible, but 1 2 and 3 is possible with time. Help? Everything is possible with time :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip registration fails
I have this problem for 2 days and i dont understand
I am behind a nat
my sip.conf is:
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-sip
disallow = all
allow= gsm
allow= ilbc
allow= ulaw
allow= alaw
;
;
localnet = 172.27.254.0/255.255.255.0 ; intern network ip address
;localmask = 255.255.255.0 ;
externip =193.49.116.12 ; my public ip address
;
maxexpirey=180
defaultexpirey=160
;
register = 560793:[EMAIL PROTECTED]/6002
;
[fwd]
type=friend
secret=mypasswd
username=fayafibun
host=fwd.pulver.com
fromdomain=fwd.pulver.com
insecure=very
context = from-sip
;
;
;
;
[bombaclaat]
callerid=(bombaclaat 6009)
type=friend
secret=mypasswd
host=dynamic
auth=md5
defaultip=172.27.254.14
context=internal
reinvite=no
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=all
mailbox=bombaclaat
qualify=1000
nat=yes
;
;
[6002]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
disallow=all
allow=all
;context=internal
context = from-sip
mailbox=6002
;
[6000]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
disallow=all
allow=all
context=internal
mailbox=6000
;
[bloodclaat]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
disallow=all
allow=all
context=internal
mailbox=bloodclaat
;
;
my extension.conf
[general]
static=yes
writeprotect=no
[globals]
;
; The name to use on callerid
;
BOMBA=SIP/bombaclaat
OTRE=SIP/6002
FWDUSERID=560793
FWDUSERNAME=fayafibun
PHONE1=6002
PHONE1VM=voicemail(6002)
FWDEXTEND=6002
;EVRYONE=${BOMBA}${OTRE}
;
[internal]
;
; local extensions
;
exten = bombaclaat,1,Dial(SIP/bombaclaat,60) ; call SIP extension
bombaclaat for 60 seconds, if extension bombaclaat is called
exten = bombaclaat,2,Voicemail(ubombaclaat) ; if we cant connect
to bombaclaat or after seconds go to the unavail VM
exten = bombaclaat,102,Voicemail(bbombaclaat); if busy, go to the busy VM
exten = 6002,1,Dial(SIP/6002,60) ; call SIP extension
bombaclaat for 60 seconds, if extension bombaclaat is called
exten = 6002,2,Voicemail(u6002) ; if we cant connect
to bombaclaat or after seconds go to the unavail VM
exten = 6002,102,Voicemail(b6002); if busy, go to the busy VM
exten = bloodclaat,1,Dial(SIP/bloodclaat,60)
exten = bloodclaat,2,Voicemail(ubloodclaat)
exten = bloodclaat,103,Voicemail(bbloodclaat)
exten = 6000,1,Dial(SIP/6000,60)
exten = 6000,2,Voicemail(u6000)
exten = 6000,103,Voicemail(b6000)
exten = _[123456789],1,NoOp(callfor${EXTEN})
exten = _[123456789],2,Dial(SIP/${EXTEN},40,tr)
exten = _[123456789],3,Congestion
exten = 1312605133,1,Dial(${FIPC}/${EXTEN:1},60) ; call SIP
extension bombaclaat for 60 seconds, if extensio$
exten = 1312605133,2,Voicemail(ubombaclaat) ; if we cant connect
to bombaclaat or after seconds go to t$
exten = bombaclaat,104,Voicemail(bbombaclaat);;
;
;appeler le 2500 de n importe kel phone pour contacter le voicemail system
exten = 2500,1,VoicemailMain
exten = 2500,2,Hangup
;
;
; Voicemail System
;
exten = 123,1,Answer
exten = 123,2,Playback(tt-weasels)
exten = 123,3,Voicemail(6002)
exten = 123,4,Hangup
;
;
;exten = ,1,VoiceMailMain(${CALLERIDNUM}) ; extension is
the VM system,
; go directly to callers VM
;exten = ,2,Hangup
;
;[outbound-internal]
;
; include local extensions
;
; include = internal
;
;
; include SIP accounts
;
; include = 6002
; include = bombaclaat
; include = 6000
; include = bloodclaat
[default]
;
; include from-sip for default. We dont use it, but it might be a good idea
;
;include = internal
;Extension Description
;101 Mark Spencer
;102 Wil Meadows
;0 Operator
include = from-sip
include = fwd-out
[fwd-out]
exten = _7.,1,SetCIDNum(${FWDUSERID})
exten = _7.,2,SetCIDName(${FWDUSERNAME})
exten = _7.,3,Dial(SIP/fwd-outgoin/${EXTEN:1})
exten = _7.,4,Playback(invalid)
exten = _7.,5,Hangup
[from-sip]
exten = ${FWDEXTEN},1,Dial(${PHONE1},30)
exten = ${FWDEXTEN},2,Voicemail(u${PHONE1VM})
exten = ${FWDEXTEN},3,Hangup
exten = ${FWDEXTEN},102,Voicemail(b${PHONE1VM})
exten = ${FWDEXTEN},103,Hangup
I have those errors
Jan 20 11:30:18 NOTICE[98310]: chan_sip.c:4053 sip_reg_timeout:
Registration for '[EMAIL PROTECTED]' timed out, trying again
Jan 20 11:30:24
[Asterisk-Users] Asterisk - libunicall - MFCr2 *** settings problems ***
Dear Steve and *.* e1r2 developers and users, now MFCR2 is successfully installed! many thanks for your help. I'm living in Argelia. I have configure my MFCR2 according argentina R2 settigs. (look at the end of the message) the testcall run perfectly (only warnings and I think that is just debug). but I have many problems and when I run Asterisk-MFCR2, generally in the begging no errors occures. after random time many inopportune errors occures: sound-cuts, dumb intervals, drop calls and disconnections!!! I think that R2 settings I use are not adapted to Argelian R2 settings. I will try change them but I have not idea where I must do changes. Please help me. Regards, kaws P.S: in the following: my configuration - Argelian R2 parameters and an example of error. *** my configuration is : unical.conf: protocolclass=mfcr2 protocolvariant=ar,20,4 protocolend=cpe span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 loadzone=us defaultzone=us zaptel.conf: span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 * the R2 setting of argelia according to quintum are : CD-Bits ::: 0001 Invert-Bits ::: DNIS Length ::: 9 digits * Answer Tone ::: A-6 Send 1st Digit 1 Group B Xmt Idle Tone : B-6 Group B Xmt Busy Tone : B-3 Group B Rcv Idle Tones B-2 B-3 Group B Rcv Busy Tones B-1 B-2 ANI Request ::: Do not request ANI ANI Length ANI Category Request Tone ANI Tone Request ANI Category :: I-1 ANI Calling Party Category II-1 Seizure Ack Timeout ::: 150ms Release Guard Timeout : 600ms * Always double-check the DNIS length with your carrier *** example of inopportune disconnection: Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Rx bits 0x9 [1/ 20/103/107] Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Far end disconnected - state 0x20 Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:2548 handle_uc_event: UC event Far end disconnected Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:2854 handle_uc_event: Far disconnect cause 16 Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:1088 unicall_hangup: unicall_hangup(UniCall/5-1) Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:1263 unicall_hangup: Causes 0 16 Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 mfcr2_DropCall() Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Call disconnected - state 0x800 Jan 17 10:04:17 WARNING[-1114432592]: chan_unicall.c:2548 handle_uc_event: UC event Drop call Jan 17 10:04:17 WARNING[-1114432592]: chan_unicall.c:2892 handle_uc_event: Doing a uc_ReleaseCall Jan 17 10:04:17 WARNING[-1114432592]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 mfcr2_ReleaseCall() Jan 17 10:04:17 WARNING[-1114432592]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Tx bits 0x9 [1/1000/106/107] Jan 17 10:04:17 WARNING[-1114432592]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Destroying call with CRN 32769 Jan 17 10:04:17 WARNING[-1114432592]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 release_guard_expired Jan 17 10:04:17 WARNING[-1114432592]: chan_unicall.c:2548 handle_uc_event: UC event Release call __ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ztdummy and meetme conference problem
HiA followup on my previous problem (no sound) description: I compiled zaptel and ztdummy, and loaded themThen i recompiled asterisk and configured sip clients and a conference. When i load the ztdummy module into the kernel, and run asterisk, the conference room seems to work, but i cannot hear any communication between clients, or even the demo (extension 1000)if i unload the ztdummy, and run asterisk again, the demo is audible, but conference room stops working.Everything is running on Whitebox Linux respin 1, kernel 2.4.21-15.EL on a VMware workstation 4.5.2I am using X-lite softphones to test the setup.There is no firewall between the server and the clients (they are in the same LAN)Please HelpBozhidar___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] change domain caller
Hello. I want to change the domain in the from url when making a call. I can change de user ID with SetCallerId but asterisk adds @192.168.1.2 How can I define what to add to the CallerID in extensions.conf? Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] hardware details
I'd say you'd need at least a Quad Pentium 4 Xeon 3.2ghz server. 4gb of RAM and a Raid 5 array with 5 120gb 10,000 rpm SCSI drives. Check ibm.com for it. :-) Just Kidding. Seriously though, 8 extensions isn't much. When you say trunk lines though, do you mean 2 Voice T1/PRI's? Each with 23 phone lines? That's 46 phone lines for 8 extensions. Maybe I'm misreading it, it is 5am here. For only 8 extensions, the server can be as simple as a P3 500mhz, w/ 512mb of RAM and a 20gb hard drive. Make sure linux is cleanly installed and only needed services are loaded. You can get this system off of PriceWatch.com or something. The analog conversion can be done by 4 Sipura SPA-2000's, or SPA-2100's if you need router capabilities. I personally use Dell PowerEdge rack mount servers, like the 1650 or the 1750 models. I also use Gentoo Linux, my understanding is BKW (a well-know asterisk contributor) uses a similar setup. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, January 20, 2005 5:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] hardware details Hello, I want to build a PBX with the following specs : 1. we have two trunk lines 2. we need upto 8 extensions - all analog phones maybe one voip phone What is hardware that I need and where to find it ? Thanks Varun ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls
Are you using PC or mac? Isamar On Thu, 20 Jan 2005, Wilson Pickett wrote: I would also suggest that while it is possible to do something, it is not always wise :) See the significant volumes of reports in the archives regarding multiple zaptel cards in one system. I must be lucky: I have 2 X100P and a TDM400 with zero IRQ or other issues. And double NAT for the voIP part. :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queue log analyser?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ben Merrills wrote: | There's a few (open source/free) ones in development. I myself am | developing one of them. | | Ben | Hi. Why not join all the project in just one ? Actually which queue log analyzers projects are beeing developed ? Check the mail from Ben Merrills sent to the list 14-10-2004 15:10. I don't know if he releases the source code, but, from the screenshots it seems to be a good one. Jo?o Amaro - -- Begin Mail | I've been doing some work on a queue log analyser for a while now, | getting the basics in place, an example of which you can find at | the URL below. However, just wondering what information people | think is most useful in a log analyser? | | At present it includes the following features: | | # Time periods - specify a period of days from the log which you | want to generate statistics for (e.g. only the last 14 days) # | Templating - allows the stats to be inserted into any html/text | template using specific tags to insert stats. This means you could | create a number of templates and execute the analyser against them | to give different information on different pages (quite flexible). | # Specify start and end dates - similar to the first feature, | except you can specify a tight period from your log, not just the | last x number of days # Channels/Agents to names - simple text file | allows you to specify a name, agent number and a channel - e.g. | Ben, Agent/1, Sip/ben. This is then used in the output # instead | of raw data # JPG graphs - includes a custom class to generate line | graphs of information (e.g. hourly call volumes etc) | | What I want to know though is, what output people would like. At | the moment there is an overview of all queues, which includes: | | Total Calls, total connected calls, total abandoned calls, calls | abandoned within x seconds, calls exited with key press, Average | hold time, max hold time, average talk time | | Agent overview includes: Calls taken, Average talk time | | Graph of call volume per hour of the day Graph of call volume per | day (over the period specified) | | Runs under windows (.NET or mono required) or any other OS that | support .NET/mono (Linux, Mac, BSD etc) | | http://muad.xdev.net/Projects/qig/sample.html | | | Not really done anything like this before, so as much input as | possible would be appreciated. | | Cheers, | | Ben -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFB75EfJUm/Bor63CERAv/UAKCwiYZ96RLqX0m7Ks9eL1f7iG4IDQCcCWvK gafg+vLAgQpjl75Hp5y8tug= =PwR8 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] queue log analyser?
I've not released the source yet, I asked last week on the mailing list for people to send me over some example queue_logs, because so far I've only been able to test the software against my own. I have however made a lot of changes to it since last I posted about it. Template engine has been improved Allows for recursion of a directory of templates Allows for different output directories (so you can do a daily, weekly and monthly all from the same set of templates say) And quite a few other bits As soon as I get some sample data that people don't mind the results being posted for then I can show it off a bit more. Hope to get some sample data soon, Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of João Amaro Sent: 20 January 2005 11:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] queue log analyser? -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ben Merrills wrote: | There's a few (open source/free) ones in development. I myself am | developing one of them. | | Ben | Hi. Why not join all the project in just one ? Actually which queue log analyzers projects are beeing developed ? Check the mail from Ben Merrills sent to the list 14-10-2004 15:10. I don't know if he releases the source code, but, from the screenshots it seems to be a good one. Jo?o Amaro - -- Begin Mail | I've been doing some work on a queue log analyser for a while now, | getting the basics in place, an example of which you can find at | the URL below. However, just wondering what information people | think is most useful in a log analyser? | | At present it includes the following features: | | # Time periods - specify a period of days from the log which you | want to generate statistics for (e.g. only the last 14 days) # | Templating - allows the stats to be inserted into any html/text | template using specific tags to insert stats. This means you could | create a number of templates and execute the analyser against them | to give different information on different pages (quite flexible). | # Specify start and end dates - similar to the first feature, | except you can specify a tight period from your log, not just the | last x number of days # Channels/Agents to names - simple text file | allows you to specify a name, agent number and a channel - e.g. | Ben, Agent/1, Sip/ben. This is then used in the output # instead | of raw data # JPG graphs - includes a custom class to generate line | graphs of information (e.g. hourly call volumes etc) | | What I want to know though is, what output people would like. At | the moment there is an overview of all queues, which includes: | | Total Calls, total connected calls, total abandoned calls, calls | abandoned within x seconds, calls exited with key press, Average | hold time, max hold time, average talk time | | Agent overview includes: Calls taken, Average talk time | | Graph of call volume per hour of the day Graph of call volume per | day (over the period specified) | | Runs under windows (.NET or mono required) or any other OS that | support .NET/mono (Linux, Mac, BSD etc) | | http://muad.xdev.net/Projects/qig/sample.html | | | Not really done anything like this before, so as much input as | possible would be appreciated. | | Cheers, | | Ben -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFB75EfJUm/Bor63CERAv/UAKCwiYZ96RLqX0m7Ks9eL1f7iG4IDQCcCWvK gafg+vLAgQpjl75Hp5y8tug= =PwR8 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ilbc high bandwidth
Good day all We have 2 asterisk servers connected to each other via IAX2 using ilbc. Each call we make goes up to 25kbit and each one there after 25kbit as well Is there a way to bring it down? Pleas Help Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] monitoring packet loss?
Hi Is it possible to somehow monitor/log packet loss and/or jitter in RTP? I want to know how things look if someone complains about audio. Best regards roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Operator Panels?
It's called asternic, www.asternic.org .. The client is based on flash which connects to a perl daemon on the server. It uses the manager (manager.conf) interface to determine extension status. Pretty neat :-) Matt -Original Message- From: David John Walsh [mailto:[EMAIL PROTECTED] Sent: Thursday, January 20, 2005 12:22 AM To: Nicolás Gudiño; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Operator Panels? On 20 Jan 2005, at 03:06, Nicolás Gudiño wrote: Hello, The problem we're having is transfers don't seem to work? ie: when someone calls inbound, you drag and drop the call on the extension you'd like and it just bridges the 2 phones together instead of transfering the call? Maybe this was intentional or maybe I'm just doing something wrong? Other than that the panel seems to work great. You can set reverse_transfer to 1 in op_server.cfg and it will transfer the other leg of the call (Ex: if you drag phone A to phone B, it will transfer the other leg of phone A (maybe an iax trunk or whatever) to B, instead of dropping the trunk and bridging A with B. I am interested in the product that is being described here, but have only recently joined the discussion list. I guess my question is in 2 parts : a) what is the product / area of asterisk that is being refered to within this email b) is there an archive of messages for this reflector that I can browse before posting questions? Kind regards David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Park/retrieval of calls
Hi! I'd like to be able to park/pickup a call with h323. Parking the call is easy, using ParkAndAnnounce. But ParkAndAnnounce does not return the parkinglot in a variable. So, I can't retrieve the call later. To keep it simple, here is a simple scenario of what I want: The call is established from phone 307. I enter DTMF code #1 on phone 307. The call is then parked. I hangup. I dial a number (#1 for exemple) on phone 307 to retrieve the call. Call must be successfully retrieved. Alternatively, I can dial #1307 from another phone to retrieve the call parked from phone 307. Maybe there is another (better way) to perform this. Mickal Ciss. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Transfers.
Am Mittwoch, 19. Januar 2005 19:18 schrieb Asterisk List: The current CVS HEAD version already has ## transfer built-in. See the included configs/features.conf.sample file. You can define your own transfer key sequence. There is also an attended transfer feature. What is an attended transfer? :) -- Robert Spielmann - TAL.DE Klaus Internet Service GmbH [EMAIL PROTECTED] Robertstr. 6 * D-42107 Wuppertal, Germany Tel +49 (0) 202 495-364 * Fax +49 (0) 202 / 495-399 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk from flash with dynamic voicemail enable/disable?
I would like to create a rock solid asterisk server that comes up no matter what. Ofcourse I can consider hardware raid1 but for the cost of a hardware raid controller I can also buy a 2GB compact flash card that doesn't produce any heat or noise and is friendly to the electricity bill and our environment :) 2GB is more than sufficient for a linux distro and an * installation but 512 Mb may not be enough for voicemail for 40 users. For this reason I am considering to use a small (notebook) harddrive but this will create another point of failure. (Alternatively I can consider an NFS share) Is there any way to make asterisk automatically disable all voicemail but continue running if there is any problem with the voicemail partition, disk or NFS share? Thanks for any hints / tips etc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some more hardware and E1 questions
Hi again folks! ;) As before, I will transform one E1 30 Channel PRI into 30 FXS channels using Adit 600. Now I'm into choosing server platform. And the two opponents are: * Dell PowerEdge 750 w/ SCSI RAID (or even SATA RAID1) * FujistuSiemens PRIMERGY RX100 S2 (SATA RAID1) As I've seen people having problem with HP server, I havn't looked at it at all. What experience do you have with the alternatives above? Which would you recommend? And another question at the same time; what's really E1? How is E1 devices connected? Seems like regular Cat5 cables, but it problably ian't? If anyone's using Adit 600, did they send all cables required for connecting to the FXS channels? Seems like a very unique plug on the side of Adit. Thanks! BR Daniel Nyström ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to manage Digium TDM04B outgoing calls correctly
So if you think the server can handle 5 TDM400P cards let me know. I've done an installation with 5 TDM400P cards - 4 PSTN lines and 12 analog phones. There are no outstanding issues that havent been solved by tweaking a particular config option (e.g echo, callprogress issues etc...). Gerald. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] latest cvs will not compile
Good day, I just downloaded the latest CVS and it will not compile. This is the error I receive: pbx_dundi.c:54:18: zlib.h: No such file or directory pbx_dundi.c: In function `update_key': pbx_dundi.c:1313: warning: implicit declaration of function `crc32' pbx_dundi.c: In function `dundi_decrypt': pbx_dundi.c:1369: warning: implicit declaration of function `uncompress' pbx_dundi.c:1369: `Z_OK' undeclared (first use in this function) pbx_dundi.c:1369: (Each undeclared identifier is reported only once pbx_dundi.c:1369: for each function it appears in.) pbx_dundi.c: In function `dundi_encrypt': pbx_dundi.c:1394: warning: implicit declaration of function `compress' pbx_dundi.c:1395: `Z_OK' undeclared (first use in this function) make[1]: *** [pbx_dundi.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/pbx' make: *** [subdirs] Error 1 What do I need to do? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Media Path Optimization NAT
Let me restate my problem. I have a group of users behind a constrained pipe to the public network. There are a few mobile users that will mostly be working from their home offices. I *really* want to avoid having a call from a mobile user to a public number cause double the traffic on the corporate link. Am I making any kind of sense? You're making sense, but trying to use the canreinvite=yes is not going to be the answer in my opinion. As stated previously, for that to work as you'd like, the sip provider would need to initiate the reinvite and its certainly not in their best interest to do that (not to mention the time they would consume trying to make it work with unknown nat functions at your user's multiple locations). There are lots of other ways to address the issue, but in my opinion each approach will require spending additional funds. You really need to identify the different ways to handle the requirement and the costs associated with each. Don't know of any way around that. Sorry to be a bother, but other ways to you see to address the issue? I'm certainly willing to invest time and funds into this, that isn't an issue. Is SER really the solution to having greater control over the SIP transactions and their associated RTP streams? I'm not a SER user, therefore others on this list might have a better understanding as to its appropriateness. Other possible approaches: - two * systems, one of which is colocated outside your corp structure with iax link, and a sip client with two proxy registration definitions (for internal system, if sip client isn't registered, send call to colocated system) - two sip accounts; one internal and one with a sip provider, sip client with two different registrations, dialplan to support both - second internet pipe at your corp location dedicated to outbound calls to your sip provider (iax-gsm across broadband?) - existing config but use a lower-bandwidth codec and increase the size of your broadband pipe to support required bandwidth - two broadband pipes; one for basic internet use, second dedicated only to * (remote sip client registration and calls via sip provider). If * configured with registered IP, sip client only needs one registration Obviously, having a good understanding as to the maximum number of simultanous calls (to your sip provider) is needed to size pipes, etc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to manage Digium TDM04B outgoing calls correctly
On January 20, 2005 11:42 am, Begumisa Gerald M wrote: I've done an installation with 5 TDM400P cards - 4 PSTN lines and 12 analog phones. I'm curious -- what is the motherboard you're doing this on? CPU? That's a lot of interrupt load! -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_Capi initial deadlock
Hello, I am using chan_capi 0.3.5 and Asterisk CVS-v1-0-12/29/04-15:32:48 on a SuSE Linux 9.0 with Kernel 2.4.21-99-default In the system is a AVM C4 with one port connected to PSTN at PTP BRI and another one to an ISDN PBX with an PMP BRI. The system is running fine, but I have regualary this error, and then chan_capi is not working anymore. Jan 18 15:29:46 WARNING[2919]: Avoided initial deadlock for 'CAPI[contr1/1429092]/128', 10 retries! Jan 18 16:00:09 WARNING[2919]: Avoided initial deadlock for 'CAPI[contr1/1429092]/128', 10 retries! I have searched the archives and found two hints: 1.) Hard disk 2.) Patch to chan_capi To 1.) I do not think that is the problem. It is an older PIII 500 but with only 5 users there should not be a problem? top shows 92 to 98 % processor idle time. To 2.) I did not tried it. The patch should solute that problems and enable faxing? Has anybody experiences with it? If there is a problem why is not kapejod solving that? I hope you could help me, I have some really angry customers. Regards Felix ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] :: Success Case = Motorola 62802-51 as FXO device ::
Hello list , I´d like to report a success case with a modem based on chipset : Motorola 62802-51. It works fine , and zaptel identifies as a X100P ( not clone ) . Red Alarms can be identified . :) This doesn´t occurred on MD3200 ambient chipsets. can you send us more info? driver,versions,logs, audio experience (echo, delay, ...) --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Stress Test
Title: Message Hi, Is there a free toll for SIP stress testing that supports RTP? Can SIPp be used for such purposes (to send audio)? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk - libunicall - MFCr2 *** settings problems***
Kaws: Are you using unicall-0.0.1d or earlier? If yes, please switch to 0.0.2pre4 (or pre3 if you have sound problems with pre4) and test again. 0.0.1 versions had a lot of problems, mostly in outgoing calls Guillermo From: kaws elchamal [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-dev@lists.digium.com, asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk - libunicall - MFCr2 *** settings problems*** Date: Thu, 20 Jan 2005 02:37:38 -0800 (PST) MIME-Version: 1.0 Received: from lists.digium.com ([69.16.138.164]) by mc5-f33.hotmail.com with Microsoft SMTPSVC(6.0.3790.211); Thu, 20 Jan 2005 02:49:11 -0800 Received: from [69.16.138.164] (localhost [127.0.0.1])by lists.digium.com (Postfix) with ESMTPid 3024D2FEDAB; Thu, 20 Jan 2005 04:37:49 -0600 (CST) Received: from psmtp.com (exprod5mx124.postini.com [64.18.0.38])by lists.digium.com (Postfix) with SMTP id B91AD2FC342for asterisk-users@lists.digium.com;Thu, 20 Jan 2005 04:37:30 -0600 (CST) Received: from source ([66.218.93.197]) by exprod5mx124.postini.com([64.18.4.10]) with SMTP; Thu, 20 Jan 2005 04:37:38 CST Received: (qmail 66919 invoked by uid 60001); 20 Jan 2005 10:37:38 - Received: from [82.101.157.10] by web42104.mail.yahoo.com via HTTP;Thu, 20 Jan 2005 02:37:38 PST X-Message-Info: 820stLNiepS16Rm4VQXrmlJV/GNqC7nB3m+8dRPKmTA= X-Original-To: asterisk-users@lists.digium.com Delivered-To: asterisk-users@lists.digium.com Comment: DomainKeys? See http://antispam.yahoo.com/domainkeys DomainKey-Signature: a=rsa-sha1; q=dns; c=nofws; s=s1024; d=yahoo.com;b=3ykjJ3afigUOYwBsx96OOddfq2B6XuphFSm22oYD3lR4o7p4zCD3RR+t5VcLcABRHa4WxOZtnpSrvTv6eSan9OXFOuCOpVDCETawEQwvg7KusVjRxQ+mTrD0vxSH9tlDiN5FK2YZLLe8MregamaHyuQOjKa29PTNqXZ84kzJlU0=; X-pstn-levels: (S:49.76371/99.9 ) X-pstn-settings: 1 (0.1500:0.1500) gt3 gt2 gt1 X-pstn-addresses: from [EMAIL PROTECTED] [65/3] X-BeenThere: asterisk-users@lists.digium.com X-Mailman-Version: 2.1.5 Precedence: list List-Id: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users.lists.digium.com List-Unsubscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] List-Archive: http://lists.digium.com/pipermail/asterisk-users List-Post: mailto:asterisk-users@lists.digium.com List-Help: mailto:[EMAIL PROTECTED] List-Subscribe: http://lists.digium.com/mailman/listinfo/asterisk-users,mailto:[EMAIL PROTECTED] Errors-To: [EMAIL PROTECTED] Return-Path: [EMAIL PROTECTED] X-OriginalArrivalTime: 20 Jan 2005 10:49:12.0659 (UTC) FILETIME=[AD93D230:01C4FEDD] Dear Steve and *.* e1r2 developers and users, now MFCR2 is successfully installed! many thanks for your help. I'm living in Argelia. I have configure my MFCR2 according argentina R2 settigs. (look at the end of the message) the testcall run perfectly (only warnings and I think that is just debug). but I have many problems and when I run Asterisk-MFCR2, generally in the begging no errors occures. after random time many inopportune errors occures: sound-cuts, dumb intervals, drop calls and disconnections!!! I think that R2 settings I use are not adapted to Argelian R2 settings. I will try change them but I have not idea where I must do changes. Please help me. Regards, kaws P.S: in the following: my configuration - Argelian R2 parameters and an example of error. *** my configuration is : unical.conf: protocolclass=mfcr2 protocolvariant=ar,20,4 protocolend=cpe span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 loadzone=us defaultzone=us zaptel.conf: span=1,1,0,cas,hdb3 cas=1-15:1101 cas=17-31:1101 * the R2 setting of argelia according to quintum are : CD-Bits ::: 0001 Invert-Bits ::: DNIS Length ::: 9 digits * Answer Tone ::: A-6 Send 1st Digit 1 Group B Xmt Idle Tone : B-6 Group B Xmt Busy Tone : B-3 Group B Rcv Idle Tones B-2 B-3 Group B Rcv Busy Tones B-1 B-2 ANI Request ::: Do not request ANI ANI Length ANI Category Request Tone ANI Tone Request ANI Category :: I-1 ANI Calling Party Category II-1 Seizure Ack Timeout ::: 150ms Release Guard Timeout : 600ms * Always double-check the DNIS length with your carrier *** example of inopportune disconnection: Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Rx bits 0x9 [1/ 20/103/107] Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:634 unicall_error: UniCall: mfcr2 Far end disconnected - state 0x20 Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:2548 handle_uc_event: UC event Far end disconnected Jan 17 10:04:17 WARNING[-1120736336]: chan_unicall.c:2854 handle_uc_event: Far disconnect cause 16 Jan 17 10:04:17 WARNING[-1120736336]:
Re: [Asterisk-Users] ilbc high bandwidth
Good day all We have 2 asterisk servers connected to each other via IAX2 using ilbc. Each call we make goes up to 25kbit and each one there after 25kbit as well bit rate is 1bps, giving 1667 bytes/sec packetization is 20ms, giving 34 bytes per packet IAX header is 4 bytes UDP header is 8 bytes IP header is 20 bytes this means one packet is 34+4+8+20=66 bytes 50 packets per second gives 3300 bytes/per second, meaning 26400bps Is there a way to bring it down? yes hack asterisk to use a lower packetization value roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls
Then if let say instead of buying TDM400P cards I get this : Clipcomm CG-410 Quad FXO Gateway is it any good? They also sell Quad FXS Gateway. Clipcomm seems to sell the cheapest Quad FXO/FXS Gateways arround so I'm wondering if it's working fine with asterisk. I found this one too but at a lot higher price : AudioCodes MP108 8-Port FXO Analog Gateway (SIP) I need to buy a conference phone also and I'm looking at Cisco or Polycom. Anyone tested one of the 2? Last concern about making my channels in a group and add that group in my dial plan. How can I make sure it will start with channel 4 and not pick a random one between the 3 channels as I'm pretty sure if I put in my dial plan a group having channel 2, 3 and 4 it might do the opposite and start with channel 2 then if it's busy switch to 3 and then 4 instead of 4 then 3 then 2 no? Thanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Webmin Module for Asterisk
There is already one, you can find it here : ftp://ftp.asterisk.org/pub/asterisk/webmin But I never managed to make it work, maybe it should be updated Anybody wanna take the challenge ? :) BTW, I've done some web pages that show you your configuration, and let you edit the text files in your browser. If you want it, drop me a message ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1x fxs + 1x fxo transfer
hi, i have 1 PSTN line and ip or analog phone i need get call(with phone ip or analog) from PSTN and transfer it(i.e. to sales) to the asterisk on corporate network pstn - gw - asterisk | phone can you recommend me some hardware (cheapest than PC+fxo card+asterisk)? --- Marek Cervenka Centrum Vypocetni Techniky CVT - http://cvt.fpf.slu.cz FPF SLU OPAVA - http://www.fpf.slu.cz LCNA- http://lcna.slu.cz === ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls
Last concern about making my channels in a group and add that group in my dial plan. How can I make sure it will start with channel 4 and not pick a random one between the 3 channels as I'm pretty sure if I put in my dial plan a group having channel 2, 3 and 4 it might do the opposite and start with channel 2 then if it's busy switch to 3 and then 4 instead of 4 then 3 then 2 no? I think * start with 1, then 2, ... until it finds an available channel. I you really want it to start with 4, then 3 ... I think just re-managing your lines so that you primary number (line 1) is plugged in port 4, and vice-versa, then put all those lines in the same group, and tell * to dial by this group, it would solve your problem. If I'm wrong, please correct me ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls
In extensions.conf the order you list the channels for a given dial plan does not matter, the priority you set for the channel is the order that the system utilizes. Can't help you with the other questions. I use Digium T1 cards to a channel banks. John Dunham -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Martin Roy Sent: Thursday, January 20, 2005 3:00 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls Then if let say instead of buying TDM400P cards I get this : Clipcomm CG-410 Quad FXO Gateway is it any good? They also sell Quad FXS Gateway. Clipcomm seems to sell the cheapest Quad FXO/FXS Gateways arround so I'm wondering if it's working fine with asterisk. I found this one too but at a lot higher price : AudioCodes MP108 8-Port FXO Analog Gateway (SIP) I need to buy a conference phone also and I'm looking at Cisco or Polycom. Anyone tested one of the 2? Last concern about making my channels in a group and add that group in my dial plan. How can I make sure it will start with channel 4 and not pick a random one between the 3 channels as I'm pretty sure if I put in my dial plan a group having channel 2, 3 and 4 it might do the opposite and start with channel 2 then if it's busy switch to 3 and then 4 instead of 4 then 3 then 2 no? Thanks Martin Roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial plan problems with realtime extensions ...
Hi, Case1: - -- extensions.conf exten = 1023,1,Voicemail(101) exten = 1023/101,1,MeetMe(200) Case2: - - extensions table (using realtime extensions) ++-+--++--+-+ | id | context | exten|priority| app | appdata | ++-+--++--+-+ | 29 | default | 1023 |1 | Voicemail | 101| | 30 | default | 1023/101 |1 | MeetMe| 200| In the first case when user 101 dials 1023, it directs him to meetme room 200. But in the case of realtime extensions it directs user 101 to Voicemail of 101, like any other user. It doesn't consider 1023/101 entry. How can I achieve proper routing in case of realtime ? Thanks, ~Vamsi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Operator Panels?
Use lists.digium.com for list browsing On Thu, 20 Jan 2005 06:47:53 -0600, Matt Schulte [EMAIL PROTECTED] wrote: It's called asternic, www.asternic.org .. The client is based on flash which connects to a perl daemon on the server. It uses the manager (manager.conf) interface to determine extension status. Pretty neat :-) Matt -Original Message- From: David John Walsh [mailto:[EMAIL PROTECTED] Sent: Thursday, January 20, 2005 12:22 AM To: Nicolás Gudiño; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Operator Panels? On 20 Jan 2005, at 03:06, Nicolás Gudiño wrote: Hello, The problem we're having is transfers don't seem to work? ie: when someone calls inbound, you drag and drop the call on the extension you'd like and it just bridges the 2 phones together instead of transfering the call? Maybe this was intentional or maybe I'm just doing something wrong? Other than that the panel seems to work great. You can set reverse_transfer to 1 in op_server.cfg and it will transfer the other leg of the call (Ex: if you drag phone A to phone B, it will transfer the other leg of phone A (maybe an iax trunk or whatever) to B, instead of dropping the trunk and bridging A with B. I am interested in the product that is being described here, but have only recently joined the discussion list. I guess my question is in 2 parts : a) what is the product / area of asterisk that is being refered to within this email b) is there an archive of messages for this reflector that I can browse before posting questions? Kind regards David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 1x fxs + 1x fxo transfer
pstn - gw - asterisk | phone can you recommend me some hardware (cheapest than PC+fxo card+asterisk)? That's the kind of stuff the sipura 3k really shines at. It offers one FXS, one FXO, 2 VoIP channels and decent routing capabilities for about $100. I've never managed to get echo quite right with X100P. On the other hand, the sipura unit Just Works. Combined with VoIP capabilities of Asterisk, it makes a really solid combination - I love it! iCanDream I would _love_ to see a unit that has the same build quality as the sipura 3000 work with a mini embedded asterisk, asterisk no-frills clean dial plan syntax, and IAX2 support. Add a little salt of network auto-discovery logic, and you would get a dream ATA and a killer for plug-and-playability thanks to Asterisk's superior protocol IAX2. /iCanDream Cheers, Jean-Michel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Stress Test
SIPp has no facility to originate audio/media, it can just send back the media it receives on its RTP port, more like an RTP proxy. ~Vamsi On Thu, 20 Jan 2005 14:55:20 +0100, Stojan Sljivic - Pamet [EMAIL PROTECTED] wrote: Hi, Is there a free toll for SIP stress testing that supports RTP? Can SIPp be used for such purposes (to send audio)? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Call-Waiting
On Thu, Jan 20, 2005 at 01:16:42PM +1100, Adam Goryachev said: On Wed, 2005-01-19 at 10:43 -0500, C F wrote: On Wed, 19 Jan 2005 15:44:44 +1100, Adam Goryachev [EMAIL PROTECTED] wrote: On Tue, 2005-01-18 at 20:03 -0600, Eric Rees wrote: Has anyone been able to find a way to disable call-waiting on Polycom phones? I've not yet found any solution to this, and I haven't seen anyone else who has. Definitely please let us all know if you do find the answer... http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup Fixing lazy top-posting. Good call (bad pun intended... :-) setgroup doesn't work in all cases. Consider that the user may be receiving calls from methods other than the dialplan (eg, queues) I haven't thought it through, but I'll throw this idea into the wind... If you route all calls through an extension macro (inbound and outbound,) could you have an asterisk DB variable that is set/reset when a line is in use? I take it ChanIsAvail will return true if one call is already in progress which is why we can't use it... In addition, this call macro could add / remove extensions from a queue when a call is in progress... I have no idea what the impact would be if you did something like transfer a call... Sure would be nice if Polycom pulled their head out of their *$$ and started supporting their product properly. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Stress Test
Hi, Is there any other free tool for SIP testing that has facility to originate audio/media? Regards, Stojan Sljivic -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vamsi Pottangi Sent: Thursday, January 20, 2005 15:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Stress Test SIPp has no facility to originate audio/media, it can just send back the media it receives on its RTP port, more like an RTP proxy. ~Vamsi On Thu, 20 Jan 2005 14:55:20 +0100, Stojan Sljivic - Pamet [EMAIL PROTECTED] wrote: Hi, Is there a free toll for SIP stress testing that supports RTP? Can SIPp be used for such purposes (to send audio)? Regards, Stojan Sljivic ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ilbc high bandwidth
Roy Sigurd Karlsbakk wrote: Good day all We have 2 asterisk servers connected to each other via IAX2 using ilbc. Each call we make goes up to 25kbit and each one there after 25kbit as well bit rate is 1bps, giving 1667 bytes/sec packetization is 20ms, giving 34 bytes per packet Actually, iLBC in asterisk uses 30ms frames.. IAX header is 4 bytes UDP header is 8 bytes IP header is 20 bytes you're also forgetting the ethernet, PPP, or other low-level overhead.. this means one packet is 34+4+8+20=66 bytes 50 packets per second gives 3300 bytes/per second, meaning 26400bps Is there a way to bring it down? yes hack asterisk to use a lower packetization value Or use trunk mode, which can do this for single calls (try setting trunkfreq to 60), and also significantly reduces overhead for multiple calls.. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] G.729? Worth it?
Andrew Kohlsmith wrote: On January 19, 2005 12:23 pm, Paul Fielding wrote: I think you might want to clarify that Best audio quality is in relation to other highly compressed codecs. Certainly my (albeit limited) experience is that g711 is much more clear than g729. Compared against gsm, for example, however, the audio quality is quite good heh -- everyone keeps bashing on GSM but out of the low bitrate codecs I've tried (G711, GSM, iLBC) GSM is king (G711 is the absolute upper bound of my low bitrate) Since when is g711 low bitrate? it to sound good. Any ideas for additional testing would be great -- I'm not afraid of packet captures or code hacking but I'm not sure where to begin at this point. My links are solid (no packet loss, low jitter, you name it) and as I said... G711, GSM, ulaw... these all sound great. It's just iLBC. -A. We briefly tested iLBC and found that the audio quality was not acceptable. If you have the ability, you may also want to try out Speex. Other than the high CPU overhead, many people here have found that it gives you good audio quality while using less bandwidth than g711 ulaw. One of the biggest problems with Speex is finding good phones that support it. Our clients mainly use Polycom and Cisco phoes, which do not support Speex. -- Aaron Johnson Star Networks Main: 602-889-3000 | Office: 602-889-3002 | Cell: 602-741-4660 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Operator Panels?
I couldn't find this option, I'm running the latest stable there is an unstable version, is it in that one? -Original Message- From: Nicolás Gudiño [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 19, 2005 9:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Operator Panels? Hello, The problem we're having is transfers don't seem to work? ie: when someone calls inbound, you drag and drop the call on the extension you'd like and it just bridges the 2 phones together instead of transfering the call? Maybe this was intentional or maybe I'm just doing something wrong? Other than that the panel seems to work great. You can set reverse_transfer to 1 in op_server.cfg and it will transfer the other leg of the call (Ex: if you drag phone A to phone B, it will transfer the other leg of phone A (maybe an iax trunk or whatever) to B, instead of dropping the trunk and bridging A with B. -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan_Capi initial deadlock
On Thursday, 20 January, 2005 14:42 : Felix Deierlein
[EMAIL PROTECTED] wrote:
Jan 18 16:00:09 WARNING[2919]: Avoided initial deadlock for
'CAPI[contr1/1429092]/128', 10 retries!
2.) Patch to chan_capi
I did not tried it. The patch should solute that problems and enable
faxing? Has anybody experiences with it? If there is a problem why is
not kapejod solving that?
You should try :)
If you don't want the fax support, you can just change this line :
--- original/chan_capi.c Fri Aug 13 12:07:28 2004
+++ chan_capi/chan_capi.c Wed Oct 27 18:55:32 2004
@@ -556,7 +556,7 @@
}
}
// wait for the B3 layer to go down
- while (i-state != CAPI_STATE_CONNECTED) {
+ while ((i-state != CAPI_STATE_CONNECTED) (i-state !=
CAPI_STATE_DISCONNECTED)) {
usleep(1);
}
}
kapejod is (was ?) quite unresponsive.
--
Carl
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Re: [Asterisk-Users] E911 Testing !
Joe Greco wrote: 911 Testing is a very complicated issue. For a clec it typically involves scheduling with them so they will expect your call. Also we frequently use false addresses (that are MSAG resolvable) and some very sophisticated PSAPs even have fake addresses that MSAG resolve to a testing ESN. Translated in english: 1. I put in a special address mapped to a phone number into the 911 location database. This is in the ALI database. The primary source of data that the 911 centers map phone number to address. 2. MSAG (The master street address guide) maps actual street addresses to ESNs an ESN is an Emergency Service Number (or something like that, feel free to correct me). It is basically a specific collection of Police, Fire and EMS. For example, Your house might use Police A, Fire B and EMS B, but the people on the other side of the street might use Police C, Fire B, EMS B (maybe it's jurisdictionally a different town). The PSAPs make up a fake address like 1234 Network Testing Blvd and they make it resolve to ESN 555 which will route to a testing center (joe) who only recieves test calls. Ok.. so too much information.. right? Definitely. Unless you happen to be doing a CLEC's office, none of it has any bearing on the original question. :-) here's the short answer. Please don't call 911 unless you have an emergency. False. Local policies vary widely. Our 911 service here in Milwaukee is the preferred method for reporting debris on the freeway to the Sheriff's Department, for example - a dispatcher once scolded me for *not* calling 911, though admittedly this was only a few years after a truck dropped some debris on I-94 that ultimately punctured the gas tank of a minivan containing a large family and lots of people died, so people have been more sensitive to debris on the highway. In fact, around here, it's fairly common for installers to test 911 service, because there's a danger in 911 *not* working as advertised under ordinary conditions (someone forgot this or that, not too hard on a PRI). Find out who your local PSAP is and call the administative number for it and talk to them. Sometimes it is hard to find this number, but it's out there. Look for Emergency services in ACME town or ACME Town 911 Dispatch etc,etc. Some very small towns actually have their administrative lines forward to the 911 centers for those areas. Call the police department's non-emergency number and they can help track down who to contact, if all else fails. Also be aware that if you are a carrier, you are required by law to have a signed contract with the 911 agency. This is typically so they can collect on the federally mandated 911 end user line fees. Most offices aren't phone carriers. Even most offices for carriers won't have an installer putting in phones that knows anything about some contract locked up half a dozen states away in the Legal Department vault at LEC Headquarters. So that's not too useful to the guy who just wants to verify correct operation of 911 services for an office install. The short form: *ASK* your local 911 center what they prefer you to do. In general, they *want* 911 to work right, and there will be some way to get you what you need. ... JG Ok, So maybe too much information for you. 911 is a mystery to most people and regardless if you are a carrier or not this is how it works. In short, you better make sure it works. Not just because you may be liable (if something happens, everyone gets sued, right?) but because it's the right thing to do(tm). You *want* 911 to work. Really. Now some areas are perfectly happy with you just casually dialing 911 and making sure it works. Sure they want it to work too. But this is **highly** dependent on what area you are in. Everyone has their own policy. I personally would never start out by trying to call 911 and seeing how they react. Calling your police department's non-emergency number may be a very good way to start off. Many (most) large cities have rules about when testing can be done. Houston for example don't do any testing on Mondays or Fridays or on Weekends, and other days testing can only be done until 2pm. Also, they don't like to test if it is raining or other unusual weather. And for the most part, these rules make a lot of sense. BTW whoever your provider is (assuming you are *not* a LEC) can probably give you some insight as how to test 911.. Even if you are a simple POTS customer. -Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using Zyxel Analog Telephone adapter with a GSM gateway
Searching through wiki and google.
http://www.2n.cz/products/gsm_gateways/voip/voiceblue.html
but there are also other products on the
market.
---
Wondering if its possible to connect as
follows:
Extension - Asterisk -
ZyxelAnalogTelephoneAdapter - GSM gateway.
The best way would be to make the ZyxelAnalog.. to
be a channel.But I don't think that is doable.. or ?
So i checked with Dial command.. trying to
use something like:
exten =
99,1,Dial(SIP/11,20,D($EXTEN),w=800ms)
; Dial option
; 'D([digits])'
-- Send DTMF digit string *after* called party has
answered;
but before the bridge. (w=500ms sec pause)
Problem is, that the astrisk won't push the $EXTEN
numbers.
(or does it ? I can't hear anything :-|
)
My console(verbose level 3):
-- Executing Dial("SIP/03-031e",
"SIP/11|20|D(987654321)|w=200ms") in new stack -- Called
11 -- SIP/11-1644 is ringing --
SIP/11-1644 answered SIP/03-031e -- Attempting native
bridge of SIP/03-031e and SIP/11-1644Jan 20 16:27:43 WARNING[10949]:
chan_sip.c:1820 sip_write: Asked to transmit frame type 64, while native formats
is 4 (read/write = 4/4)Jan 20 16:27:43 WARNING[10949]: chan_sip.c:1820
sip_write: Asked to transmit frame type 64, while native formats is 4
(read/write = 4/4)...and so on for 10 lines..
then I hang up. == Spawn extension (wx3trunk, 99, 1) exited non-zero
on 'SIP/03-031e'
/ Stig Henning
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Re: [Asterisk-Users] Asterisk from flash with dynamic voicemail enable/disable?
You *could* write a script to check the voicemail partition and slap this in a cronjob. If it finds a problem, then have it switch out your extensions.conf with one that is the exact same except it plays a voicemail is currently unavailable-type message when you either try to check your vmail or are redirected to it. On Thu, 20 Jan 2005 14:13:33 +0100 (CET) Remco Barende [EMAIL PROTECTED] wrote: I would like to create a rock solid asterisk server that comes up no matter what. Ofcourse I can consider hardware raid1 but for the cost of a hardware raid controller I can also buy a 2GB compact flash card that doesn't produce any heat or noise and is friendly to the electricity bill and our environment :) 2GB is more than sufficient for a linux distro and an * installation but 512 Mb may not be enough for voicemail for 40 users. For this reason I am considering to use a small (notebook) harddrive but this will create another point of failure. (Alternatively I can consider an NFS share) Is there any way to make asterisk automatically disable all voicemail but continue running if there is any problem with the voicemail partition, disk or NFS share? Thanks for any hints / tips etc. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Richard Kolkovich [EMAIL PROTECTED] Team Leader LinuxForums.org Content Development ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] monitoring packet loss?
Roy Sigurd Karlsbakk wrote: Hi Is it possible to somehow monitor/log packet loss and/or jitter in RTP? I want to know how things look if someone complains about audio. ethereal can do some of this for rtp, I think. At the very least, if the endpoint supports RTCP (most do, except for asterisk), it can show you the contents of the RTCP RRs, which should contain this information. Getting this stuff into asterisk would be in bug 2532, bug 2863, and bug 3236. [and not just getting stats, but actually improving quality under these conditions]. -SteveK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Advanced Agents - Need a nice web interface
http://bugs.digium.com/bug_view_page.php?bug_id=0003252 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Asterisk 1.0.3 startup
Just for future reference, I think that the ldd command might have helped you figure out where files are that are being looked for. For example, on my system: aslan:/home/dana# ldd /usr/sbin/asterisk libdl.so.2 = /lib/libdl.so.2 (0x40017000) libpthread.so.0 = /lib/libpthread.so.0 (0x4001a000) libncurses.so.5 = /lib/libncurses.so.5 (0x4002f000) libm.so.6 = /lib/libm.so.6 (0x4006d000) libresolv.so.2 = /lib/libresolv.so.2 (0x4008e000) libssl.so.0.9.6 = /usr/lib/libssl.so.0.9.6 (0x4009e000) libc.so.6 = /lib/libc.so.6 (0x400cb000) libcrypto.so.0.9.6 = /usr/lib/libcrypto.so.0.9.6 (0x401e8000) /lib/ld-linux.so.2 = /lib/ld-linux.so.2 (0x4000) aslan:/home/dana# Just a tip for anyone who didn't know about that command. Maybe it's useless to you all. I don't know. __ Dana Olson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nic le Roux Sent: Thursday, January 20, 2005 4:24 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] FW: Asterisk 1.0.3 startup Sorry all, Did that and its going good now. Rgds Nic From: Nic le Roux [mailto:[EMAIL PROTECTED] Sent: 20 January 2005 11:22 AM To: 'asterisk-users@lists.digium.com' Subject: Asterisk 1.0.3 startup Hi All, I've managed to compile make and make install asterisk on Mandrake 9.2. However on startup I get the following message: [cdr_tds.so]Jan 20 11:13:54 WARNING[20999]: loader.c:258 ast_load_resource: libtds.so.3: cannot open shared object file: No such file or directory Jan 20 11:13:54 WARNING[20999]: loader.c:440 load_modules: Loading module cdr_tds.so failed! I have freetds installed from RPM, and the lib is here /usr/local/lib/libtds.so.3.0.0 Where does asterisk look for the lib ? Maybe I can do a symlink ? Any help appreciated. Thanks and Regards Nic Disclaimer: The information transmitted in this message is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination, or other use of or taking of any action in reliance upon this information by persons or entities other than the intended recipient is prohibited. If you received this message in error, please contact the sender and delete the material from any system. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: how to manage Digium TDM04B outgoing calls
On Thu, 2005-01-20 at 09:18, [EMAIL PROTECTED] wrote: Last concern about making my channels in a group and add that group in my dial plan. How can I make sure it will start with channel 4 and not pick a random one between the 3 channels as I'm pretty sure if I put in my dial plan a group having channel 2, 3 and 4 it might do the opposite and start with channel 2 then if it's busy switch to 3 and then 4 instead of 4 then 3 then 2 no? I think * start with 1, then 2, ... until it finds an available channel. I you really want it to start with 4, then 3 ... I think just re-managing your lines so that you primary number (line 1) is plugged in port 4, and vice-versa, then put all those lines in the same group, and tell * to dial by this group, it would solve your problem. If I'm wrong, please correct me ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users What about the G vs g setting for hunt criteria when using groups for outdial? d ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Troubles with Broadvoice (register)
Sometimes I have problems and changing to another of their servers makes it start working again. There probably is a way to make * deal with this properly. I am using the broadvoice account for test purposes at this time so I just edit sip.conf and restart * when this happens. What I have observed is that the server I can't register with will still have good ping times when this happens. Helder Rogério [MICROREDE] wrote: Hi! Are you also getting in trouble while trying to register in Broadvoice? Cumprimentos / Best regards, Helder Rogério __ Microrede - Tecnologias de Informação, Ltd. http://www.microrede.pt *** « There are only two types of people in the world, those who have lost data and those who will. » -- Richard Nixon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tips do update Asterisk and AMP
Hi all. Somebody knows if AMP will work with the newest version of asterisk(1.0.3)!? Somehting that I need to know before update!? How is the best way to get my system updated!? Thanks. Denis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Troubles with Broadvoice (register)
Hi! But the only server they gave for sip registration is sip.broadvoice.com I have several for outbound proxy proxy.chi.broadvoice.com and etc... Do you have any other for sip? Best regards, Helder - Original Message - From: Paul [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, January 20, 2005 4:15 PM Subject: Re: [Asterisk-Users] Troubles with Broadvoice (register) Sometimes I have problems and changing to another of their servers makes it start working again. There probably is a way to make * deal with this properly. I am using the broadvoice account for test purposes at this time so I just edit sip.conf and restart * when this happens. What I have observed is that the server I can't register with will still have good ping times when this happens. Helder Rogério [MICROREDE] wrote: Hi! Are you also getting in trouble while trying to register in Broadvoice? Cumprimentos / Best regards, Helder Rogério __ Microrede - Tecnologias de Informação, Ltd. http://www.microrede.pt *** « There are only two types of people in the world, those who have lost data and those who will. » -- Richard Nixon ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Webmin Module for Asterisk {Scanned}
I will try out your pages.. Thanks, David PS I would love to work on your Asterisk Webmin pages, but I don't know how. On Thu, 2005-01-20 at 06:01, [EMAIL PROTECTED] wrote: There is already one, you can find it here : ftp://ftp.asterisk.org/pub/asterisk/webmin But I never managed to make it work, maybe it should be updated Anybody wanna take the challenge ? :) BTW, I've done some web pages that show you your configuration, and let you edit the text files in your browser. If you want it, drop me a message ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Weird Zaphfc - not dialling non-local numbers
Hi all, I really hope that you guys can help, because I've been tearing my hair out for the past 5 hours on this one. I have a Zaphfc (BRI) card in TE mode connected to the S-Bus of a Nortel Meridian phone system. Phone calls from the Nortel to say MSN 510 are correctly being sent to the right SIP phone. When asterisk dials say Zap/g2/224 (a Nortel internal extension) the call goes through, no problem. The wierd bit is, when Asterisk calls Zap/g2/907748xx to reach an mobile on an outside line, the call connot be connected. I know for a fact that the '9' prefix is valid for use on the S-Bus, because I previously used AVM Fritz (CAPI) card on the same S-Bus. If any can help, I will be eternally grateful. Thanks, John *** Log *** voip*CLI -- Executing Dial(Zap/1-1, Zap/g2/228) in new stack -- Called g2/228 -- Accepting call from '224' to '521' on channel 0/1, span 1 -- Zap/4-1 is ringing received TEI check request for TEI = 127 received TEI check request for TEI = 127 -- Channel 0/1, span 1 got hangup -- I hung up Jan 20 16:17:26 WARNING[4613]: app_dial.c:369 wait_for_answer: Unable to forward frame -- Hungup 'Zap/4-1' == Spawn extension (from-isdn, 521, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/g2/907748xx) in new stack -- Called g2/907748xx -- Accepting call from '224' to '518' on channel 0/1, span 1 received TEI check request for TEI = 127 received TEI check request for TEI = 127 received TEI check request for TEI = 127 -- Channel 0/1, span 2 got hangup -- Hungup 'Zap/4-1' == No one is available to answer at this time -- Executing VoiceMail2(Zap/1-1, u100) in new stack -- Playing 'voicemail/default/100/unavail' (language 'en') *** extensions.conf *** [from-isdn] exten = 518,1,Dial(Zap/g2/907748xx) exten = 521,1,Dial(Zap/g2/228) *** zapata.conf *** [channels] language=en rxwink=300 switchtype=euroisdn nationalprefix= - also tried 0 and 90 here internationalprefix= - tried 00 here signalling = bri_cpe_ptmp prilocaldialplan=local pridialplan=local rxgain=3.0 txgain=3.0 echocancel=64 echotraining=yes echocancelwhenbridged=yes immediate=no overlapdial=no group = 1 context=from-isdn channel = 1-2 signalling = bri_cpe_ptmp pridialplan=local prilocaldialplan=local rxgain=3.0 txgain=3.0 echocancel=64 echotraining=yes echocancelwhenbridged=yes immediate=no overlapdial=no group = 2 context=from-isdn channel = 4-5 signalling = bri_cpe_ptmp prilocaldialplan=local pridialplan=local rxgain=3.0 txgain=3.0 echocancel=64 echotraining=yes echocancelwhenbridged=yes immediate=no overlapdial=no group = 3 context=from-isdn channel = 7-8 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SHORELINE IP100
Yes, the Shoreline IP100 is just a rebranded Polycom Soundpoint IP500, and uses the same software. By default it uses a MGCP image, but it can be changed to run SIP. See http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones for more info. B. J. -Original Message- From: Robert Augustyn [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 19, 2005 23:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SHORELINE IP100 Hi, Is it the same as IP500? Does it run the same software or do I need to flash it? Is so whare do I get it? Thanks a lot. robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Meetme Limitations?
Has anyone testing the maximum limitation for people in a Meetme conference? -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: E911 Testing ! {Scanned}
As a Firefighter I would call the 911 Dispatch Center first using there office number (XXX-). Tell them what you would like to do, what time, address and phone number. Then give them you cell. If something happens with the test they will call you back on your cell. Also some Dispatch centers can run the same test on one of there business lines. David On Wed, 2005-01-19 at 14:21, Jason Kawakami wrote: -Original Message- I believe the 911 is a serious issue if one does an asterisk installation in an office. How do you test 911? Won't they arrest you or something for dialing 911 for no reason and talking to one of their agents who could have taken a more important call? -speaking from 10+ years of installations, dial 911 and tell the operator your name, who you are with, and that you are testing a new phone system. Confirm with them that the telephone number and address they have in their system is correct, say thank you and hang-up. On occasion, you get a surly operator who has had a bad day but crap, if you had their job, your days may not be so good either. On the other hand what an emergency comes up (like someone got seriously injured) and on top of that asterisk crashed all of a sudden bringing the whole office PBX down. Since it would be not be possible to place a call and emergency matter becomes more serious, who would be held responsible? The person who installed the PBX for not implementing a redundant and reliable system? -document that on 'X' date and 'Y' time, you tested and confirmed that 911 access was functioning and have the client sign off on the installation. After that, the system is theirs. Always test emergency services access for premises equipment based solutions unless you have signed documentation from the client that they do not want 911 access out of their system! Jason Kawakami www.optellabs.com Salt Lake City, UT ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Transfers.
Attended transfer, also called supervised transfer, works like this: While on conversation with another party, you dial ** the transfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hang up and you will be back to your original conversation. On Thu, 20 Jan 2005 13:59:36 +0100, Robert Spielmann [EMAIL PROTECTED] wrote: What is an attended transfer? :) -- Robert Spielmann --JJL44 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tips do update Asterisk and AMP
Hi all. Somebody knows if AMP will work with the newest version of asterisk(1.0.3)!? Somehting that I need to know before update!? How is the best way to get my system updated!? Thanks. Denis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BRI Fax out through PRI?
Having seen all the discussions about troubles with faxing I am thinking of another solution, keeping it all digital. I want to put two ISDN BRI cards in a faxserver (not running Linux). The two BRI ports will be connected to a QUAD BRI card in an Asterisk box. The asterisk box will have a single PRI span for communication to the PSTN. Basically the setup will be: Faxserver BRI - QuadBRI in * box - PSTN through PRI in Asterisk box By keeping it digital all the way I'm hoping to avoid echo and connection problems and keep the connection speeds high. Anyone ever tried such a setup? Thanks!! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SHORELINE IP100
I believe that there is a Sip software version 1.3.4 available for that phone. The freedomphone.com has only 1.3.1. Any idea where to get the latest? Thanks a lot for your help. robert --- B. J. Bomar [EMAIL PROTECTED] wrote: Yes, the Shoreline IP100 is just a rebranded Polycom Soundpoint IP500, and uses the same software. By default it uses a MGCP image, but it can be changed to run SIP. See http://www.voip-info.org/tiki-index.php?page=Polycom%20Phones for more info. B. J. -Original Message- From: Robert Augustyn [mailto:[EMAIL PROTECTED] Sent: Wednesday, January 19, 2005 23:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SHORELINE IP100 Hi, Is it the same as IP500? Does it run the same software or do I need to flash it? Is so whare do I get it? Thanks a lot. robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SHORELINE IP100
Robert Augustyn wrote: Hi, Is it the same as IP500? Does it run the same software or do I need to flash it? Is so whare do I get it? Thanks a lot. robert It is a Polycom IP500 running MGCP image if you're using ShoreTel. We just finished a major ShoreTel installation at my work place. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Weird Zaphfc - not dialling non-local numbers
_Update_ Strangely, I've just found that I can dial local (6 digit) numbers using the '9' prefix - Zap/g2/9742xxx. I'm probably doing something really daft, but for the life of me, I can't see how I managed to create this strange scenario. John John McEleney wrote: Hi all, I really hope that you guys can help, because I've been tearing my hair out for the past 5 hours on this one. I have a Zaphfc (BRI) card in TE mode connected to the S-Bus of a Nortel Meridian phone system. Phone calls from the Nortel to say MSN 510 are correctly being sent to the right SIP phone. When asterisk dials say Zap/g2/224 (a Nortel internal extension) the call goes through, no problem. The wierd bit is, when Asterisk calls Zap/g2/907748xx to reach an mobile on an outside line, the call connot be connected. I know for a fact that the '9' prefix is valid for use on the S-Bus, because I previously used AVM Fritz (CAPI) card on the same S-Bus. If any can help, I will be eternally grateful. Thanks, John *** Log *** voip*CLI -- Executing Dial(Zap/1-1, Zap/g2/228) in new stack -- Called g2/228 -- Accepting call from '224' to '521' on channel 0/1, span 1 -- Zap/4-1 is ringing received TEI check request for TEI = 127 received TEI check request for TEI = 127 -- Channel 0/1, span 1 got hangup -- I hung up Jan 20 16:17:26 WARNING[4613]: app_dial.c:369 wait_for_answer: Unable to forward frame -- Hungup 'Zap/4-1' == Spawn extension (from-isdn, 521, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/g2/907748xx) in new stack -- Called g2/907748xx -- Accepting call from '224' to '518' on channel 0/1, span 1 received TEI check request for TEI = 127 received TEI check request for TEI = 127 received TEI check request for TEI = 127 -- Channel 0/1, span 2 got hangup -- Hungup 'Zap/4-1' == No one is available to answer at this time -- Executing VoiceMail2(Zap/1-1, u100) in new stack -- Playing 'voicemail/default/100/unavail' (language 'en') *** extensions.conf *** [from-isdn] exten = 518,1,Dial(Zap/g2/907748xx) exten = 521,1,Dial(Zap/g2/228) *** zapata.conf *** [channels] language=en rxwink=300 switchtype=euroisdn nationalprefix= - also tried 0 and 90 here internationalprefix= - tried 00 here signalling = bri_cpe_ptmp prilocaldialplan=local pridialplan=local rxgain=3.0 txgain=3.0 echocancel=64 echotraining=yes echocancelwhenbridged=yes immediate=no overlapdial=no group = 1 context=from-isdn channel = 1-2 signalling = bri_cpe_ptmp pridialplan=local prilocaldialplan=local rxgain=3.0 txgain=3.0 echocancel=64 echotraining=yes echocancelwhenbridged=yes immediate=no overlapdial=no group = 2 context=from-isdn channel = 4-5 signalling = bri_cpe_ptmp prilocaldialplan=local pridialplan=local rxgain=3.0 txgain=3.0 echocancel=64 echotraining=yes echocancelwhenbridged=yes immediate=no overlapdial=no group = 3 context=from-isdn channel = 7-8 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tips do update Asterisk and AMP
Sorry about the repost. I got an error in the first one. Denis. Em Qui 20 Jan 2005 14:48, Denis Galvão - iSolve escreveu: Hi all. Somebody knows if AMP will work with the newest version of asterisk(1.0.3)!? Somehting that I need to know before update!? How is the best way to get my system updated!? Thanks. Denis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] softphone
Does someone know a free SIP softphone which can be used from a web page and with Asterisk? Thanks in advance ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sipura SPA-2000
does sipura support analog fax machine (14400 bps) or analog modems? the cisco ata-186 does support fax machine 9600bps anyone with a linksys pap2? thx Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Transfers.
Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 20 Jan 2005, Asterisk List wrote: Attended transfer, also called supervised transfer, works like this: While on conversation with another party, you dial ** the transfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hang up and you will be back to your original conversation. On Thu, 20 Jan 2005 13:59:36 +0100, Robert Spielmann [EMAIL PROTECTED] wrote: What is an attended transfer? :) -- Robert Spielmann --JJL44 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005-01-20%5Ce78d2d987a5e46cca50a486612386c7fC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie question - can't get Asterisk to pick up incoming call
Okay, I'm going to preface this by saying I'm sure I've overlooked something really basic here. I just need someone to hit me with a clue stick and point out what I'm missing. I've got a TDM card with four FXO modules. I've plugged one of them into a PSTN line. I'm working through the examples in the Asterisk Documentation Project guide, but I can't get Asterisk to answer the line. I don't get any diagnostics on the Asterisk console when the line rings (should I?) Here are my config files: /etc/zaptel.conf: fxsls=1-4 loadzone=us defaultzone=us /etc/asterisk/zapata.conf: [channels] language=en context=default switchtype=national signalling=fxs_ls channel = 1-4 /etc/asterisk/extensions.conf: [default] exten = s,1,Answer() exten = s,2,Playback(goodbye) exten = s,3,Hangup() The zaptel and wctdm modules are loaded. ztcfg -vv gives this output: Zaptel Configuration == Channel map: Channel 01: FXS Loopstart (Default) (Slaves: 01) Channel 02: FXS Loopstart (Default) (Slaves: 02) Channel 03: FXS Loopstart (Default) (Slaves: 03) Channel 04: FXS Loopstart (Default) (Slaves: 04) 4 channels configured. What am I missing, here? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] is it possible to use Zaphfc (BRI) exactly like i4l?
HI I.m working on echo on my asterisk and I'm wonder if is it possible to use Zaphfc (BRI) exactly like i4l? How to attach msn number to such card? Thanks in advance for any help. Regards, Corvin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] # Transfers.
features.conf bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bruce Komito Sent: Thursday, January 20, 2005 11:05 AM To: Asterisk List Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] # Transfers. Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Thu, 20 Jan 2005, Asterisk List wrote: Attended transfer, also called supervised transfer, works like this: While on conversation with another party, you dial ** the transfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. You dial the transferee number and talk with the transferee to introduce the call, then you can hang up and the other party will be connected with the transferee. In case the transferee does not want to answer the call, he/she simply hang up and you will be back to your original conversation. On Thu, 20 Jan 2005 13:59:36 +0100, Robert Spielmann [EMAIL PROTECTED] wrote: What is an attended transfer? :) -- Robert Spielmann --JJL44 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://nospam.wpti.net/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2005- 01-20%5Ce78d2d987a5e46cca50a486612386c7fC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Transfers.
I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED] wrote: Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Stress Test
We will put a graphical asterisk load tester online next week. ( i know i said this before, but now its really there :) zoa. signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] # Transfers.
Does this work with app_queue/chan_agent? Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk List Sent: 20 January 2005 17:28 To: Bruce Komito Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] # Transfers. I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED] wrote: Sorry if I missed the beginning of this thread, but I've never heard of the ** transfer key sequence, nor have I found a way to make it work. Would you mind, please explaining this further or pointing me to somewhere where it's documented? (I checked Wiki and Google but no joy.) Thanks Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Stress Test
We will put a graphical asterisk load tester online next week. ( i know i said this before, but now its really there :) zoa. signature.asc Description: PGP signature signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: API Call Bridge?
In article [EMAIL PROTECTED], taf taffey [EMAIL PROTECTED] wrote: -=-=-=-=-=- -=-=-=-=-=- Hi All, Does anyone know of a way to dial two different outbound numbers and bridge them together using the Asterisk API? I answered exactly that question on this list within the last two days. Should be easy to find as it's so recent. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SHORELINE IP100
Joseph, How did the insatllation go? Any problems? How do you power this units? Thanks. robert --- Joseph Finley [EMAIL PROTECTED] wrote: Robert Augustyn wrote: Hi, Is it the same as IP500? Does it run the same software or do I need to flash it? Is so whare do I get it? Thanks a lot. robert It is a Polycom IP500 running MGCP image if you're using ShoreTel. We just finished a major ShoreTel installation at my work place. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What's up with IAXTEL?
I finally got around to signing up with Iaxtel and Free World Dialup... the price was right, as far as that goes! I've gotten and placed calls via FWD just fine. But I can't seem to get registered, or stay registered, with Iaxtel. My logs show the story; at startup I see: Jan 20 07:14:44 VERBOSE[14121]: -- Registered to '65.39.205.121', who sees us as ... yada yada... As you can see, before asterisk is finished booting, I'm registered to FWD just fine and stay that way. Jan 20 07:17:37 VERBOSE[14121]: ESC[1;37;40mAsterisk Ready. ESC[0;37;40m-- Registered to '69.73.19.178', who sees us as --- more yada yada--- According to this, it takes a few minutes longer for iaxtel to kick in... but then... Jan 20 07:17:37 DEBUG[14121]: Raw Hangup 69.73.19.178:4569, src=5, dst=404 Jan 20 07:17:37 DEBUG[14121]: Raw Hangup 69.73.19.178:4569, src=5, dst=404 Jan 20 07:17:37 DEBUG[14121]: Raw Hangup 69.73.19.178:4569, src=5, dst=404 Jan 20 07:17:37 DEBUG[14121]: Raw Hangup 69.73.19.178:4569, src=5, dst=404 ... Jan 20 07:20:06 DEBUG[14121]: Raw Hangup 69.73.19.178:4569, src=3, dst=327 Jan 20 07:20:06 DEBUG[14121]: Raw Hangup 69.73.19.178:4569, src=3, dst=327 Jan 20 07:20:06 DEBUG[14121]: Raw Hangup 69.73.19.178:4569, src=3, dst=327 Jan 20 07:23:08 VERBOSE[14121]: -- Registered to '69.73.19.178', who sees us as ...yadayadayada... Jan 20 07:23:10 DEBUG[14121]: Raw Hangup 69.73.19.178:4569, src=3, dst=868 Jan 20 07:24:09 DEBUG[14121]: Raw Hangup 69.73.19.178:4569, src=5, dst=430 ... These messages repeat ad nauseum. Bunches of raw hangups followed by re-registrations galore. I can't get or place calls thru iaxtel, and iax2 show registrations doesn't usually show them as connected. What's the diff between iaxtel and FWD? Is Iaxtel's bandwidth maxed out? Should I just delete it from the list, and forget them? Are there settings I can change? Anybody have any advice? BTW... my FWD # is 544716 murf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # Transfers.
I have no idea if atxfer works with app_queue/chan_agent. Can anyone try it? Best regards, --JJL44 On Thu, 20 Jan 2005 17:38:25 -, Ben Merrills [EMAIL PROTECTED] wrote: Does this work with app_queue/chan_agent? Cheers, Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Asterisk List Sent: 20 January 2005 17:28 To: Bruce Komito Cc: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] # Transfers. I justed edited the Wiki Asterisk config file features.conf for this attended transfer features. Please check Wiki again for details. Best regards, --JJL44 -- --JJL44 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP-to-TDM processing on-card?
Are there any cards that work with * that do the VoIP-to-TDM processing on the cards, with multiple interfaces? The QuickNet Internet LineJack meets the description I believe, but it only has a single FXS or FXO. Are there any cards that have more than one FXS? Thanks. __ Dana Olson Disclaimer: The information transmitted in this message is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient of this message is prohibited. If you receive this message in error, please contact the sender and delete the material from any system. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Versatel PRA in Belgium/Netherlands
Michael, could you provide me with contact information for your versatel account manager or dutch versatel PRI tech person i could contact? Joachim. Michael Devenijn wrote: Problem solved : The reason was quite simple ... but annoying : Interrupts !!! damned !!! Thank you -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] namens Florian Overkamp Verzonden: wo 19/01/2005 10:08 Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion' CC: Onderwerp: RE: [Asterisk-Users] Versatel PRA in Belgium/Netherlands Hi, -Original Message- Did somebody already configured a Digium card on the network of Versatel in Belgium or the netherlands, and would like to share his configuration. (zaptel.conf / zapata.conf) We have HDLC errors (timings i presume) Yes, we have such setups. Please contact me off-list with some more info about what card you are using etc. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: VoIP-to-TDM processing on-card?
Sorry. I don't know what I'm smoking today. We need T1 interfaces... :P So let me rephrase the question: Are there any cards that work with * that do the VoIP-to-TDM processing on the cards, with multiple T1interfaces? The QuickNet Internet LineJackseems to meet the description I believe, but it only has a single FXS or FXO. Are there any cards that have multiple T1 ports? Thanks. __ Dana Olson HelpDesk Technician TELESPECTRUM, INC. 1-800-704-9111 -Original Message-From: Olson, Dana Sent: Thursday, January 20, 2005 12:54 PMTo: Asterisk Mailing List (E-mail)Subject: VoIP-to-TDM processing on-card? Are there any cards that work with * that do the VoIP-to-TDM processing on the cards, with multiple interfaces? The QuickNet Internet LineJack meets the description I believe, but it only has a single FXS or FXO. Are there any cards that have more than one FXS? Thanks. __ Dana Olson Disclaimer: The information transmitted in this message is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient of this message is prohibited. If you receive this message in error, please contact the sender and delete the material from any system. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP-to-TDM processing on-card?
First, TURN OFF HTML. On Thu, 2005-01-20 at 12:53 -0500, Olson, Dana wrote: Are there any cards that work with * that do the VoIP-to-TDM processing on the cards, with multiple interfaces? Hmm, haven't seen any cards with an ethernet and a TDM interface. You must mean a card that does the TDM and codec translation for the computer. Did you have a look at the asterisk supported hardware list? Wouldn't that seem like a very pertinent place to look? The QuickNet Internet LineJack meets the description I believe, but it only has a single FXS or FXO. Are there any cards that have more than one FXS? If you want people to obey your disclaimer, you need to read it and see how it applies to a known publicly archived mailing list. If your employer requires that disclaimer, change email addresses for use with the list or risk looking incredibly stupid every time you decide to participate on the list. __ Disclaimer: The information transmitted in this message is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient of this message is prohibited. If you receive this message in error, please contact the sender and delete the material from any system. __ -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP-to-TDM processing on-card?
Olson, Dana wrote: Are there any cards that work with * that do the VoIP-to-TDM processing on the cards, with multiple interfaces? The QuickNet Internet LineJack meets the description I believe, but it only has a single FXS or FXO. Are there any cards that have more than one FXS? It's been a long time since I've seen someone post a question to the mailing list like this. I turnip could do more research than you did. Try http://www.asteriskpbx.org/index.php?menu=hardware Try http://www.digium.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sipura SPA-2000
Mine works well with a Multimodem ZPX at 14.4kbps. I'm using a SPA-2000, the linksys should pretty much be the same deal. As I previously noted on the list, the Sipura fax settings seem to break faxing so I leave them disabled. On Thu, 20 Jan 2005 18:13:29 +0100, Sergio [EMAIL PROTECTED] wrote: does sipura support analog fax machine (14400 bps) or analog modems? the cisco ata-186 does support fax machine 9600bps anyone with a linksys pap2? thx Sergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] E911 Testing !
Thank you everyone. Makes a lot of sense... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Thursday, January 20, 2005 7:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] E911 Testing ! Joe Greco wrote: 911 Testing is a very complicated issue. For a clec it typically involves scheduling with them so they will expect your call. Also we frequently use false addresses (that are MSAG resolvable) and some very sophisticated PSAPs even have fake addresses that MSAG resolve to a testing ESN. Translated in english: 1. I put in a special address mapped to a phone number into the 911 location database. This is in the ALI database. The primary source of data that the 911 centers map phone number to address. 2. MSAG (The master street address guide) maps actual street addresses to ESNs an ESN is an Emergency Service Number (or something like that, feel free to correct me). It is basically a specific collection of Police, Fire and EMS. For example, Your house might use Police A, Fire B and EMS B, but the people on the other side of the street might use Police C, Fire B, EMS B (maybe it's jurisdictionally a different town). The PSAPs make up a fake address like 1234 Network Testing Blvd and they make it resolve to ESN 555 which will route to a testing center (joe) who only recieves test calls. Ok.. so too much information.. right? Definitely. Unless you happen to be doing a CLEC's office, none of it has any bearing on the original question. :-) here's the short answer. Please don't call 911 unless you have an emergency. False. Local policies vary widely. Our 911 service here in Milwaukee is the preferred method for reporting debris on the freeway to the Sheriff's Department, for example - a dispatcher once scolded me for *not* calling 911, though admittedly this was only a few years after a truck dropped some debris on I-94 that ultimately punctured the gas tank of a minivan containing a large family and lots of people died, so people have been more sensitive to debris on the highway. In fact, around here, it's fairly common for installers to test 911 service, because there's a danger in 911 *not* working as advertised under ordinary conditions (someone forgot this or that, not too hard on a PRI). Find out who your local PSAP is and call the administative number for it and talk to them. Sometimes it is hard to find this number, but it's out there. Look for Emergency services in ACME town or ACME Town 911 Dispatch etc,etc. Some very small towns actually have their administrative lines forward to the 911 centers for those areas. Call the police department's non-emergency number and they can help track down who to contact, if all else fails. Also be aware that if you are a carrier, you are required by law to have a signed contract with the 911 agency. This is typically so they can collect on the federally mandated 911 end user line fees. Most offices aren't phone carriers. Even most offices for carriers won't have an installer putting in phones that knows anything about some contract locked up half a dozen states away in the Legal Department vault at LEC Headquarters. So that's not too useful to the guy who just wants to verify correct operation of 911 services for an office install. The short form: *ASK* your local 911 center what they prefer you to do. In general, they *want* 911 to work right, and there will be some way to get you what you need. ... JG Ok, So maybe too much information for you. 911 is a mystery to most people and regardless if you are a carrier or not this is how it works. In short, you better make sure it works. Not just because you may be liable (if something happens, everyone gets sued, right?) but because it's the right thing to do(tm). You *want* 911 to work. Really. Now some areas are perfectly happy with you just casually dialing 911 and making sure it works. Sure they want it to work too. But this is **highly** dependent on what area you are in. Everyone has their own policy. I personally would never start out by trying to call 911 and seeing how they react. Calling your police department's non-emergency number may be a very good way to start off. Many (most) large cities have rules about when testing can be done. Houston for example don't do any testing on Mondays or Fridays or on Weekends, and other days testing can only be done until 2pm. Also, they don't like to test if it is raining or other unusual weather. And for the most part, these rules make a lot of sense. BTW whoever your provider is (assuming you are *not* a LEC) can probably give you some insight as how to test 911.. Even if you are a simple POTS customer. -Brett ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com
[Asterisk-Users] Realtime Engine
I'm going to be testing the new realtime stuff further in the next few
days, and just wanted some clarification on a couple of things before I
start on it.
I believe I can store any config file in a external config such as
mgcp.conf for example, by adding it to extconfig.conf with the below
syntax.
mgcp.conf = mysql,asterisk,mgcpchans
Doing this will require a reload of asterisk to read the changes in
(since chan_mgcp hasn't been moved to realtime yet), as well as removing
the text file called mgcp.conf, so that asterisk knows to use the
database version from extconfig.conf.
I also want to play with the extensions.conf, realtime extensions config
using the switch statement as described on the WIKI.
My question on this is if I can use the SWITCH statement in a Wildcard
match context.
like the example below.
[inbound]
exten = _XX,1,SetVar(CALLEDTO=${EXTEN})
exten = _XX,2,GotoIf($[${CALLERIDNUM} = ${EXTEN}]?3:5)
exten = _XX,3,VoiceMailMain,${EXTEN}
exten = _XX,4,Hangup
exten = _XX,5,Dial,SIP/${EXTEN}|20
exten = _XX,6,VoiceMail,u${EXTEN}
exten = _XX,7,Hangup
exten = _XX,107,VoiceMail,b${EXTEN}
exten = _XX,108,Hangup
I want to replace priority 5 with the switch statement to go to the
database and match the extension to a channel like.
[inbound]
exten = _XX,1,SetVar(CALLEDTO=${EXTEN})
exten = _XX,2,GotoIf($[${CALLERIDNUM} = ${EXTEN}]?3:5)
exten = _XX,3,VoiceMailMain,${EXTEN}
exten = _XX,4,Hangup
switch =Realtime/@
exten = _XX,6,VoiceMail,u${EXTEN}
exten = _XX,7,Hangup
exten = _XX,107,VoiceMail,b${EXTEN}
exten = _XX,108,Hangup
Then in my extensions table I will have data like the following
INSERT INTO `extensions_table` VALUES (1, 'inbound', '_5172078354',
5,'DIAL', 'SIP/5172078354');
INSERT INTO `extensions_table` VALUES (2, 'inbound', '_5172078355', 5,
'DIAL', 'SIP/5172078355');
INSERT INTO `extensions_table` VALUES (3, 'inbound', '_5172078356', 5,
'DIAL', 'SIP/5172078360');
Or do I need to put all the priorities for a specific extension in the
database for each extension?
Regards
Michale Baird
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[Asterisk-Users] SIP debugs
Other then the standard sip debug is there any other sip debug bugs like for errors, events, etc. Kurt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] E911 Testing !
Thanks, Brett, for the info! I actually /like/ the long winded descriptions. FYI - In some places, the 911 dispatchers are the same people who answer the Sheriff's, Local PD and Fire phone numbers. So, simply calling the Sheriff's Dept. and saying that you just installed a new phone system and want to test 911 would be a good place to start. Calling 911 and saying oh, I'm just testing would be a bad idea, although it happens *a lot* (older people mostly, from what I hear.) While it's a bad idea, it's better than not testing it at all. 911 dispatchers are well trained. They know how to handle all sorts of calls, including testing, info, I lost my dog and I'm dying. If they /are/ busy and you say testing they can clear the call in a matter of seconds and get back to the emergencies at hand. Obviously, if you're a CLEC, or someone who's going to be making several test calls, you'll want to establish a procedure with the dispatch center first. As other posters have pointed out, it's always far better to test (even with bad procedures) than to not test and have the system fail in an emergency. I've done volunteer work for emergency services and disaster agencies and the rule of thumb is *always*, When In Doubt, Call It In! When calling _anyone_ involved in emergency services, be brief and to the point. And, in most cases, skip introducing yourself, your company, what your working, etc. Just say what you want and answer any questions directly and briefly. ie, call the Sheriff or local PD and say I want to test 911 on my new phone system. Don't get into a long winded introduction. Also, when you're transfer to someone else (this may happen more times than you'd like). Always start by saying the same thing, I want to test 911 on my new phone system. This might sound like I'm stating the obvious, but emergency service workers are trained in effective, efficent communication. If you speak to them in the same way, you're immediately be considered professional instead of someone I have to deal with. Okay, that's my long winded post of the day. Hopefully, someone will find it useful. cheers, glenn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP-to-TDM processing on-card?
I did look there. If you read my follow up, I screwed up the original question. What I want is a card with multiple T1 ports that do the processing on the card, and not on the system CPU. Is there a mailing list for Asterisk where people treat each other in a civil manner? __ Dana Olson -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Eric Wieling Sent: Thursday, January 20, 2005 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoIP-to-TDM processing on-card? Olson, Dana wrote: Are there any cards that work with * that do the VoIP-to-TDM processing on the cards, with multiple interfaces? The QuickNet Internet LineJack meets the description I believe, but it only has a single FXS or FXO. Are there any cards that have more than one FXS? It's been a long time since I've seen someone post a question to the mailing list like this. I turnip could do more research than you did. Try http://www.asteriskpbx.org/index.php?menu=hardware Try http://www.digium.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Disclaimer: The information transmitted in this message is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination, or other use of or taking of any action in reliance upon this information by persons or entities other than the intended recipient is prohibited. If you received this message in error, please contact the sender and delete the material from any system. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
