Re: [Asterisk-Users] 503 Error
After speaking with out provider, they believe it has something to do with the silence suppression tag in the SIP headers Asterisk is sending. Is there a way to remove the silence suppression tag completely? Thanks, Doug When trying to send calls from our Asterisk PBX to our upstream termination provider, I am getting Got SIP response 503 Service Unavailable back from PROVIDER We are sending the calls without registration, there is no username and password. When we were using SER it would send them without a problem. Can someone tell me if the problem is on my side or the provider's? What would cause this problem? Thanks Doug ___ 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: Email to Fax
How are you doing it Justin? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Newman Sent: Jueves, 21 de Abril de 2005 10:53 p.m. To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Email to Fax Message: 11 Date: Thu, 21 Apr 2005 20:39:22 -0500 From: Anton Krall [EMAIL PROTECTED] Subject: [Asterisk-Users] Email to Fax Anybody doing email to fax using spandsp? Yep... Justin Newman Newman Telecom, Inc. [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 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] zap to sip caller id forwarding
Hello, I have 8 zap channels connected to my local teleco company, Id like to get callerid working on my sip phones but asterisk always sets the callerid as asterisk eventho I have callerid=asrecieved in the zapata.conf and in my dialplan I set the callerid before dialing the sip chan. Could anyone give me any pointers? -- Best regards, Tom mailto:[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] What do I need to get started?
My message to the list was definitely flame bait for my ignorance, thanks for just giving me links and a point in the right direction. In an ideal world, where insecure people wouldn't need constant validation on mailing lists to bolster their self worth, what just happened here would be more common: If someone feels like answering a question with useful info, they do. Otherwise, they move on. Unfortunately, the world is not ideal, so in this case the gods must be smiling on you. Welcome to the asterisk community :) ___ 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] LiveVoip status report
There has been improvement in the quality of LiveVoip connections. Still some packet loss and resultant choppy audio, a little worse than with Vonage or Broadvoice. As noted in several posts over the past months, they still don't handle indication of ringing on an IAX channel if the caller has dialed a number in the Asterisk switch (for instance with the DISA app). The workaround previously suggested, to Answer() and then run Ringing() doesn't work in this case, because it still sends the IAX command for ringing which LiveVoip doesn't recognize. However, Playtones(ring) does work and represents a usable workaround for the price. They claim to be working on a new session controller that will fix this and other problems. We'll see. ___ 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 increased memory
Not being a c developer, perhaps I am totally wrong, and therefore beg for your understanding ... ;) Is it normal for the * executable to be increasing in memory size ? I've noticed when using top that the * executable starts life like PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 530 root 16 0 9496 9432 4648 S 0.5 1.8 0:31 0 asterisk however, after a couple of days, the size column indicates 60MB, when there are no channels and no calls active. Is that normal ? We use ISDN pri to the outside world, and SIP (Cisco 7960) internally, with Agentcallbacklogin for queues. CVS head as of the start of the month. Julian. ___ 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] Echo cancelling with Adit 600
Do anyone have experience with echo cancelling on Adit 600? My Adit 600 consist of 5*8 FXS cards and 1 CMG Router using MGCP to Asterisk. I've turned on Echo Cancelling with 64ms as longest delay (that's maximum). But there still are great echo with delay when dialing through the telco (through an E1 and EuroISDN). Any advice will be appriciated! Thanks! -- Daniel ___ 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] Line Noise UPDATE - If you've got line noise, read this
Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using X100P cards and NOT having this problem?? Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Thursday, April 21, 2005 14:23 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Line Noise UPDATE - If you've got line noise,read this Hm...well, here's something interesting. On my previous box, neither of the cards were sharing IRQs with anything.now, both cards are on 11, along with many other things. This could very well be a problem. As far as the network goes, there is very little traffic and the switch is full duplex 100 megabit. Bandwidth is only a factor on he local lan, since asterisk dials out through a FXO card(X100P) then it doesn't go across my broadband connection. Let me know your thoughts.sorry for the verbose output. Paul [EMAIL PROTECTED] /sbin]$ ./lspci -v 00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400 AGP] Host Bridge (rev 80) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master, 66Mhz, medium devsel, latency 8 Memory at d000 (32-bit, prefetchable) [size=128M] Capabilities: available only to root 00:01.0 PCI bridge: VIA Technologies, Inc.: Unknown device b198 (prog-if 00 [Normal decode]) Flags: bus master, 66Mhz, medium devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 I/O behind bridge: 9000-9fff Memory behind bridge: e800-e9ff Prefetchable memory behind bridge: d800-e7ff Capabilities: available only to root 00:09.0 Communication controller: Tiger Jet Network Inc. Intel 537 Subsystem: Intel Corp.: Unknown device 0003 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at a000 [size=256] Memory at eb002000 (32-bit, non-prefetchable) [size=4K] Capabilities: available only to root 00:0a.0 Communication controller: Tiger Jet Network Inc. Intel 537 Subsystem: Intel Corp.: Unknown device 0003 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at a400 [size=256] Memory at eb00 (32-bit, non-prefetchable) [size=4K] Capabilities: available only to root 00:0b.0 Ethernet controller: Linksys Network Everywhere Fast Ethernet 10/100 model NC100 (rev 11) Subsystem: Linksys: Unknown device 0570 Flags: bus master, medium devsel, latency 32, IRQ 5 I/O ports at a800 [size=256] Memory at eb001000 (32-bit, non-prefetchable) [size=1K] Expansion ROM at unassigned [disabled] [size=128K] Capabilities: available only to root 00:0f.0 RAID bus controller: VIA Technologies, Inc.: Unknown device 3149 (rev 80) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at ac00 [size=8] I/O ports at b000 [size=4] I/O ports at b400 [size=8] I/O ports at b800 [size=4] I/O ports at bc00 [size=16] I/O ports at c000 [size=256] Capabilities: available only to root 00:0f.1 IDE interface: VIA Technologies, Inc. VT82C586/B/686A/B PIPC Bus Master IDE (rev 06) (prog-if 8a [Master SecP PriP]) Subsystem: VIA Technologies, Inc. VT8235 Bus Master ATA133/100/66/33 IDE Flags: bus master, medium devsel, latency 32 I/O ports at c400 [size=16] Capabilities: available only to root 00:10.0 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00 [UHCI]) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at c800 [size=32] Capabilities: available only to root 00:10.1 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00 [UHCI]) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at cc00 [size=32] Capabilities: available only to root 00:10.2 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00 [UHCI]) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at d000 [size=32] Capabilities: available only to root 00:10.3 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00 [UHCI]) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at d400 [size=32] Capabilities: available only to root 00:10.4 USB Controller: VIA
[Asterisk-Users] How to attended/supervisor transfer
Hi all I don't know how to do an attended call transfer in asterisk. Iam using asterisk1.0.7 and oh323 client ( gnomemeeting ). I have to do any dialplan for that. can anyone help me to know. Thanks Varadhan ___ 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] dialling problem with astcc
when a call comes on zap astcc.agi script launch and ask caller about his card number,and when the caller is dialing his card number(56170) sometimes astcc take it by missing a number as (5670) or doubled number as (556170) i dont know whats the problem is it from zap or is it from astcc.agi script or is it from the telephone system i dont know what to do please help. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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] One touch voicemail on Cisco 7940/60
On Thu, 2005-04-21 at 21:36, Ron Wellsted wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Morris, Simon wrote:
Hello,
I'd like to program my Cisco phones to authenticate themselves to
voicemail upon hitting the right button on my 7940/60's
Ideally the voicemail app will detect which extension the call is coming
from and drop the user straight into the menu.
Is this possible?
Many thanks
~sm
Yes this is possible.
In your extensions.conf:
exten = _8501,1,Answer()
exten = _8501,2,VoicemailMain(s${CALLERIDNUM})
exten = _8501,3,Hangup()
then program the messages button to dial 8501 either via settings, SIP
Settings, Messages URI (set to 8501) or in the SIPDefault.cnf file
However if you are busy callers will immediately be redirected to this
extension and get your voicemail menu unless you have call waiting
enabled on the phone. Suggest you try this:
; Assuming your extension is 2034 and 8501 is your voicemail extension.
;
exten = 8501/2034,1,VoicemailMain(s2034)
exten = 8501,1,Voicemail(b2034)
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[Asterisk-Users] Spandsp 0.0.2Pre15 with bristuff-0.2.0-RC8 Problem - Hangup
Hi ! After succesfully setting up a Server with an E1 Card -Asterisk CVS and Spandsp-0.0.2Pre10, I am having a problem getting the combination of bristuff-0.2.0-RC8.tar.gz and Spandsp-0.0.2pre15 to work on another machine. I have a trust HFC card I want to use. The problem described was identical on 2 linux machines, 1 Via Epia M1 and 1 Asus Celeron 400 MHz, both running Debian. My Linux is Debian Sarge with custom kernel 2.4.30, with all the libraries and linux Sourcecode (Kernel 2.4.30) installed to compile both packages (libtiff, libxml, etc.). Compiling is no problem. First I started the scripts in bristuff, which then download asterisk 1.0.7 and all the other packages needed. After installing zaphfc and successfully checking the asterisk telephony functions, I installed the spandsp lib (--prefix=/usr), and copied the apps to the asterisk directory. Compiled asterisk again, and moved the apps to the modules dir of asterisk. Restarted Asterisk -gc as root. Set Verbose to 5 , Debug to 99. Calling the extension I reserved for fax ( exten = 45,1,rxfax(/home/master/testfax.tif) I receive this: *CLI -- Executing RxFAX(Zap/1-1, /home/master/testfax.tif) in new stack Urgent handler -- Accepting voice call from '' to '45' on channel 0/1, span 1 Urgent handler Urgent handler -- Channel 0/1, span 1 got hangup Urgent handler Urgent handler Urgent handler -- Hungup 'Zap/1-1' Urgent handler So the call is immediately hung up... What am I doing wrong ? This can't be such an exotic setup ?! TIA 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] Re: TE110p - universal voltage?
In article [EMAIL PROTECTED], Craig Guy [EMAIL PROTECTED] wrote: Can anyone with a TE110p confirm that it will fit and work in both a 3.3 and 5 volt pci slot? From photos it looks to be a universal card but the digium literature makes no mention of voltage requirements. I can cofirm that it has both the 5V and 3.3V cutouts in the edge connector. I can also confirm that I've used the card successfully in a 5V slot. I haven't tried it in the 3.3V slot. 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] Provisioning lines 5 and 6 via TFTP
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thu, 21 Apr 2005, Robert Goodyear wrote: Has anyone experienced a problem provisioning lines 5 and 6 of a Cisco 7960 via a SIPx.CNF over TFTP? What I'm experiencing is that regardless of the linex_... entries in the CNF file, lines 5 and 6 show UNPROVISIONED on the phone console, despite the fact that the rest of the line provisioning fields are correctly filled on the phone. As soon as I re-enter the line name on the keypad for line 5 -- and JUST the line name because, remember, the rest of the fields are already present -- then lines 5 and 6 appear on the phone console. Weird! /rg There is a known issue with the size of the SIP*.cnf files on some versions of the firmware. The cure for this is to stripout all comments from the .cnf files. HTH - -- Ron Wellsted http://www.wellsted.org.uk [EMAIL PROTECTED] FWD: 519961 N 52.567623, W 2.137621 -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (GNU/Linux) iQEVAwUBQmi/EEtP/KMNOfRbAQKAwAf/Zk68jmdIvEKz/B9MKm4oewmuzMFncWiO Ya79Y9iCUpYr/IlvlBVU36WYZTSOoijoUeTC4gaU0iWSMzk8pd4K6WEQ50Bt1zb1 DbIR0XSvr02uyjjRLcW+f2KJ8uMV4kaySZeJDjIU9VfJSUK0qrtbdistAWER6PV8 632LG+V/csMSlgvZ7Dt00kZYWrsLncry1L4++MIYUtBZY9VbX2n4AyYbwp0VnX5p o+Onz0BthbPbEzDcpY5LpUKB9UQvw0D/j+rUEsje9HgXuGG97q9LENg3sBbM1yR4 5v0NsQGpLnli/unJFbv2oGkgUrBMow3W5qP3DbsklSYYPzUhKVRAfQ== =rTzZ -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] Provisioning lines 5 and 6 via TFTP
Can you please post your .cnf files? On 4/21/05, Robert Goodyear [EMAIL PROTECTED] wrote: Has anyone experienced a problem provisioning lines 5 and 6 of a Cisco 7960 via a SIPx.CNF over TFTP? What I'm experiencing is that regardless of the linex_... entries in the CNF file, lines 5 and 6 show UNPROVISIONED on the phone console, despite the fact that the rest of the line provisioning fields are correctly filled on the phone. As soon as I re-enter the line name on the keypad for line 5 -- and JUST the line name because, remember, the rest of the fields are already present -- then lines 5 and 6 appear on the phone console. Weird! /rg ___ 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] mpg123 won't compile, arch x86_64
try compling using 'make linux-devl' otherwise play around with formatmp3 after doing a cvs checkout asterisk-addons On 4/21/05, Michael Welter [EMAIL PROTECTED] wrote: mpg123 won't compile on my Opteron system. Doesn't seem to like the pushl and popl assembly instructions, ie.e, pushl %ebp and others. I tried changing to pushq %rbp and it compiled but wouldn't run. Do we have any assembler programmers on the list? Thanks ___ 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] X100P delayed ring on incoming calls?
I have setup an asterisk box with 3off X100P cards and hooked them up to the PSTN. So far so good, everything does what it is supposed to do for the msot part. Incoming calls seem to ring three or four times before asterisk then skips to do what it is supposed to do. If the caller drops the call before the extensions have started ringing asterisk seems not to pick this up and carries on regardless. Anyone got any ideas? This was built from CVS. == Spawn extension (BT_PSTN, s, 1) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' -- Starting simple switch on 'Zap/3-1' Apr 22 09:20:11 NOTICE[20851]: chan_zap.c:5585 ss_thread: Got event 2 (Ring/Answered)... Apr 22 09:20:13 NOTICE[20851]: chan_zap.c:5585 ss_thread: Got event 2 (Ring/Answered)... Apr 22 09:20:14 NOTICE[20851]: chan_zap.c:5585 ss_thread: Got event 2 (Ring/Answered)... Apr 22 09:20:16 NOTICE[20851]: chan_zap.c:5585 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/3-1, SIP/110SIP/112|20|tr) in new stack -- Called 110 -- Called 112 -- SIP/110-ff2f is ringing -- SIP/112-4713 is ringing -- SIP/110-ff2f answered Zap/3-1 extensions.conf [BT_PSTN] exten = s,1,Answer exten = s,2,Dial(SIP/110SIP/112,20,tr) exten = s,3,Voicemail(u000) zapata.conf context=BT_PSTN callerid=Inbound Call 01774987987 signalling = fxs_ks channel=1-3 group = 1 ___ 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] Adit 3104 - user experiences?
From what I was able to gather on CACs web site the 3104 only supports 24 analog ports. On 4/20/05, Peter Hoppe [EMAIL PROTECTED] wrote: Hello, I am looking for a solution to connect about 40 analog telephones to an Asterisk pbx. Initially I wanted to use an Adit 600 channel bank, but yesterday I talked to Carrier Access, and they recommended the Adit 3104 gateway. All I am looking for is a device that multiplexes many analog phones via one connection (preferably via ethernet). I was told that the 3104 speaks standard SIP and from what I heard it seems to be fulfilling that task. I am located in the UK. * Does anyone have any experience with this device? * Would it work with standard UK analog phones? * Are firmware upgrades free? or pay-as-you-upgrade? * Does the device use a standard SIP (i.e. fully rfc compliant)? Thank you very much! Peter Hoppe ___ 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] X100P delayed ring on incoming calls?
On Fri, 2005-04-22 at 10:22 +0100, bam wrote: I have setup an asterisk box with 3off X100P cards and hooked them up to the PSTN. So far so good, everything does what it is supposed to do for the msot part. Incoming calls seem to ring three or four times before asterisk then skips to do what it is supposed to do. If the caller drops the call before the extensions have started ringing asterisk seems not to pick this up and carries on regardless. Anyone got any ideas? This sounds like the CallerID problem. * is trying to get the ID, but the UK's method is different to the default, so it does not get an ID it finally gives up and processes the call. Look for UKCaller ID settings in the archives or Wiki. (I left the UK 12 years ago so I've never looked at it). -- Dave Cotton [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] X100P delayed ring on incoming calls?
On Friday 22 April 2005 10:45, Dave Cotton wrote: On Fri, 2005-04-22 at 10:22 +0100, bam wrote: Incoming calls seem to ring three or four times before asterisk then skips to do what it is supposed to do. If the caller drops the call before the extensions have started ringing asterisk seems not to pick this up and carries on regardless. I had this problem, and I think I tracked the problem down to the order I had the commands in my zapata.conf. Here is my working one which passes CallerID and causes * to pickup the call immediately: [channels] signalling=fxs_ks usecallerid=yes cidsignalling=v23 cidstart=usehist language=en context=from-landline echotraining=yes echocancelwhenbridged=yes echocancel=yes rxgain=1.0 txgain=-6.0 channel=1 immediate=no Previously I had the three CallerID directives *AFTER* the 'channel=1' and this seemed to confuse it.. Cheers, Gavin. ___ 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] X100P delayed ring on incoming calls?
3-4 rings seems kind of long, I usually see 1.5-2 (enough to grab the caller ID here in the States). The only way I know of to speed it up is to turn off all of the features like distinctive ring detection, caller ID, etc. -- depending on your usage, that may help some. I haven't confirmed that this actually does anything myself, but it seems logical that Asterisk could pick up quicker if it wasn't waiting for caller ID. This is what people suggest if you Google the list. Either way, the best I've ever managed on the X100P's was 1 ring before Asterisk picks up and starts doing its thing. ___ 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] X100P delayed ring on incoming calls?
On Fri, 2005-04-22 at 10:45, Dave Cotton wrote: On Fri, 2005-04-22 at 10:22 +0100, bam wrote: I have setup an asterisk box with 3off X100P cards and hooked them up to the PSTN. So far so good, everything does what it is supposed to do for the most part. Incoming calls seem to ring three or four times before asterisk then skips to do what it is supposed to do. If the caller drops the call before the extensions have started ringing asterisk seems not to pick this up and carries on regardless. Anyone got any ideas? This sounds like the CallerID problem. * is trying to get the ID, but the UK's method is different to the default, so it does not get an ID it finally gives up and processes the call. Look for UKCaller ID settings in the archives or Wiki. (I left the UK 12 years ago so I've never looked at it). What a hero, problem solved. ;-) zapata.conf usecallerid=no ___ 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] One touch voicemail on Cisco 7940/60
On Thu, 2005-04-21 at 21:36 +0100, Ron Wellsted wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Morris, Simon wrote:
Hello,
I'd like to program my Cisco phones to authenticate themselves to
voicemail upon hitting the right button on my 7940/60's
Ideally the voicemail app will detect which extension the call is
coming
from and drop the user straight into the menu.
Is this possible?
Many thanks
~sm
Yes this is possible.
In your extensions.conf:
exten = _8501,1,Answer()
exten = _8501,2,VoicemailMain(s${CALLERIDNUM})
exten = _8501,3,Hangup()
then program the messages button to dial 8501 either via settings, SIP
Settings, Messages URI (set to 8501) or in the SIPDefault.cnf file
That method works perfectly - thanks to all that took the time to answer
~sm
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[Asterisk-Users] DTFM tones almost completly muted.
Hello everyone, I am trying to receive DTMF commands on asterisk from PSTN calls terminated at my asterisk box. I have tried to terminate the PSTN calls with both SIP and IAX using sigate.co.uk and voipuser as the PSTN terminator. When I listen to tones sent from the PSTN side (e.g. continuous DTMF tone of about 3 seconds) on the asterisk server (stored in the voice mail) the tone is more or less completely muted, just the initial tone start can be heard. I am using the G711 codec. Does anyone have any idea if these tones are on purpose muted by the service providers or any other reason why it does not work? Thanks for any help. Ian Hailey. ___ 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] Dell PowerEdge SC1425 w/ TE405P?
Hello, I've been asked to build a couple of Gateway servers for a client w/ TE405P hardware, and have been looking around at various 1U options. I've been looking at SuperMicro and Tyan barbones boxes as possible platforms, but then was directed to Dell's SC1425 by a friend. Short story, is that you can purchase a 2x3.0Ghz/1GigDDR400/1xSATA box in a 1U form factor for $1,498.00. This seems almost too good to be true, so I'm asking if anyone has had any experience with this box? I'm not up on my PCI terminology, but as I understand it, the TE405P can only be used in a 32 bit 33Mhz slot at 5.0 Volts. This SC1425 lists a 1x 64-bit/1xxMHz PCI-X slot under it's expandability information. I'd venture to guess this is probably NOT going to work with a TE405P. That being said, if it works, great. If not, what 1U boxes are people using IN PRODUCTION w/ TE405P cards? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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] IAX2 Error
Anyone has any idea what does this error means when executing an IAX2 call? Apr 22 11:50:19 WARNING[9124]: Received mini frame before first full voice frame The called party can hear but the calling, no. Is this a fine tunning into iax.conf? Thanks, Robson ___ 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] livevoip callerid
I don't think it's correct to put dashes in the CIDNum.
MARK.
Paul Fielding wrote:
Hmmm... I still can't get name, though number works. Perhaps I'm
missing something?
context livevoip in iax.conf that hooks me to livevoip
dial 9 in front of long distance number to dial livevoip instead of
regular LD.
snip
LIVEVOIP=IAX2/username:[EMAIL PROTECTED]
snip
exten = _91NXXNXX,1,SetCIDNum(403-666-|a)
exten = _91NXXNXX,2,SetCIDName(Satan Lives|a)
exten = _91NXXNXX,3,Noop(Caller Name: ${CALLERIDNAME}, Number:
${CALLERIDNUM})
exten = _91NXXNXX,4,Dial(${LIVEVOIP}/${EXTEN:1})
regards,
Paul
- Original Message - From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, April 05, 2005 6:31 AM
Subject: Re: [Asterisk-Users] livevoip callerid
I'll be damned... I changed my format to match yours, and both
the SetCIDNum and SetCIDName work just fine. I could never get
the name to work properly prior to your post. Thanks!
I am able to set name and number with Livevoip. Make sure your
variables are actually being set.
exten = s,1,SetCIDNum(xx|a)
exten = s,n,SetCIDName(first last|a)
exten = s,n,Noop(Caller Name: ${CALLERIDNAME}, Number:
${CALLERIDNUM})
MARK.
Cameron Schaus wrote:
Is there any way I can send callerId information to livevoip? I have
added the following to my extensions.conf, but when I place calls
through livevoip, no callerId information is sent to the called party.
SWC_CALLERID=14031234567
SWC_CALLERNAME=foo
exten = _1NXXNXX,1,SetCallerID(${SWC_CALLERID})
exten = _1NXXNXX,2,SetCIDName(${SWC_CALLERNAME})
exten = _1NXXNXX,3,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Thanks,
Cam
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Re: [Asterisk-Users] Echo cancelling with Adit 600
On Fri, 22 Apr 2005, Daniel Nyström wrote: Do anyone have experience with echo cancelling on Adit 600? My Adit 600 consist of 5*8 FXS cards and 1 CMG Router using MGCP to Asterisk. I've turned on Echo Cancelling with 64ms as longest delay (that's maximum). But there still are great echo with delay when dialing through the telco (through an E1 and EuroISDN). I suspect a channel bank normally only cancels the echo on the extension side. To handle echo on a voip link you would need an echo canceler with several times longer span. You can try to reduce the size of any jitterbuffer in the Adit. That may make the echo latency lowe enough. 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
Re: [Asterisk-Users] DTFM tones almost completly muted.
On 22/04/05, Ian Hailey [EMAIL PROTECTED] wrote: Hello everyone, I am trying to receive DTMF commands on asterisk from PSTN calls terminated at my asterisk box. I have tried to terminate the PSTN calls with both SIP and IAX using sigate.co.uk and voipuser as the PSTN terminator. When I listen to tones sent from the PSTN side (e.g. continuous DTMF tone of about 3 seconds) on the asterisk server (stored in the voice mail) the tone is more or less completely muted, just the initial tone start can be heard. I am using the G711 codec. Does anyone have any idea if these tones are on purpose muted by the service providers or any other reason why it does not work? I'm not aware of the detailed reason, but DTMF into Asterisk from Sipgate won't work. This path is well-trodden... http://www.voipuser.org/forum_topic_844.html amongst other places. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[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] Line Noise UPDATE - If you've got line noise, read this
On Fri, Apr 22, 2005 at 02:40:10AM -0500, Paul said: Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using X100P cards and NOT having this problem?? Yes. I can make a call from a POTS phone hooked up to a Cisco ATA 186, out one X100P to the PSTN, back in a second X100P, to a phone hooked up to the second port on the ATA186 with no noise, and no echo, and a pretty small delay (which you can hear with one handset in each ear.) I have disabled most of the on-board I/O such as parallel, serial, and extra USB controllers, and the X100's are on int 5 and 7, not shared with anything. Interrupts 10 and 11 have a bunch of stuff shared and are used by USB controllers, ethernet ports (one on each IRQ) video card, SCSI controller, and one unknown device (some special nVidia device.) This machine is also used as a firewall / gateway / email server but does NOT run X (which I hear can cause problems on some machines.) I've been running this configuration for about 9 months with virtually no problems in a SOHO environment including weekly 3-hour long conference calls. I realize this doesn't help you much, but it IS possible for the configuration to work. I have been thinking about getting a Sipura 3000 to add another FXS port and remove one X100P which would also cut down on the number of interrupts, leaving me one X100P for timming (so I don't need ztdummy.) MAYBE this would help 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] asterisk@home 0.9 zap problems
-- Executing Dial(SIP/3001-e13a, ZAP/1/65869804) in new stack This is what's wrong I think. The line is missing the 'g' for the trunk group. On all of my [EMAIL PROTECTED] boxes the cli shows -- Executing Dial(SIP/227-a4dd, ZAP/g0/3428463) in new stack It depends how you set it up in AMP. Click on Setup-Trunks. Do you have a trunk named ZAP/g0 or one named ZAP/1 ? if it's ZAP/1 then click on it, and go at the bottom at Zap Identifier (trunk name) : and enter g0. Press Submit changes then apply your changes. That should fix it. hth ___ 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] X100P delayed ring on incoming calls?
Joseph Gutowski [EMAIL PROTECTED] wrote: [...] Either way, the best I've ever managed on the X100P's was 1 ring before Asterisk picks up and starts doing its thing. Well, when you think about it, it's hardly going to pick up after zero rings, is it? :) -- Beer is proof that God loves us and wants us to be happy. - Benjamin Franklin ___ 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] X100P delayed ring on incoming calls?
On Friday 22 April 2005 12:07, Peter Corlett wrote: Joseph Gutowski [EMAIL PROTECTED] wrote: [...] Either way, the best I've ever managed on the X100P's was 1 ring before Asterisk picks up and starts doing its thing. Well, when you think about it, it's hardly going to pick up after zero rings, is it? :) In the UK it's entirely possible - the CallerID info comes through as encoded data before the first ring has taken place :) Polarity change, a burst of V23 data, then the normal rings Cheers, Gavin. ___ 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] X100P delayed ring on incoming calls?
3-4 rings seems kind of long, I usually see 1.5-2 (enough to grab the caller ID here in the States). The only way I know of to speed it up is to turn off all of the features like distinctive ring detection, caller ID, etc. -- depending on your usage, that may help some. I haven't confirmed that this actually does anything myself, but it seems logical that Asterisk could pick up quicker if it wasn't waiting for caller ID. This is what people suggest if you Google the list. Either way, the best I've ever managed on the X100P's was 1 ring before Asterisk picks up and starts doing its thing. Or, try immediate=yes in zapata.conf ___ 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] X100P delayed ring on incoming calls?
Joseph Gutowski wrote: The only way I know of to speed it up is to turn off all of the features like distinctive ring detection, caller ID, etc. -- depending on your usage, that may help some. I haven't confirmed that this actually does anything myself, but it seems logical that Asterisk could pick up quicker if it wasn't waiting for caller ID. This is what people suggest if you Google the list. Either way, the best I've ever managed on the X100P's was 1 ring before Asterisk picks up and starts doing its thing. We came across this exact problem in testing, and the more we thought about it, the more we realised it's actually a non-problem. Think about it from the point of view of the person ringing. What do they hear, if there's that delay? ring ring ring ring ring ring Thank you for calling XYZ. Please press 1 for this, 2 for that hold music Hello, XYZ, Dave speaking, how can I help? Now remove the ring ring - they've barely finished dialling before they're into the Thank you for calling. I don't know about you, but I tend to use the ringing time as a bit of an opportunity to get my brain in gear. It actually *throws* me slightly if I don't hear a ring or three. If somebody's using an all-in-one type of phone, perhaps a landline with the buttons in the same physical bit you hold to your head, or a mobile, they'll probably miss the first ring or so as they move the phone from in front of them to their ear. Which is better - they miss a ring ring or they miss the initial announcement and have to go through the IVR again, or just plain get a bit flustered? Don't forget - not all callers are us. A lot of people out there still have techno-fear with things as complex as telephones. -- Adrian Chapman Director Trivas Ltd Business on the Move Mobility - Messaging - Infrastructure - Security - Remote Access 07796 690210 - 01582 626552 ___ 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] X100P delayed ring on incoming calls?
Gavin Hamill [EMAIL PROTECTED] wrote: On Friday 22 April 2005 12:07, Peter Corlett wrote: [...] In the UK it's entirely possible - the CallerID info comes through as encoded data before the first ring has taken place :) Polarity change, a burst of V23 data, then the normal rings A good point, but I gather the X100P can't detect a line inversion, so it's pretty much still got to wait until it sees loads of RICH, CHUNKY VOLTS to know there's an incoming call. -- PGP key ID E85DC776 - finger [EMAIL PROTECTED] for full key Please contribute to the beer fund and a tidier house: http://search.ebay.co.uk/_W0QQfgtpZ1QQfrppZ25QQsassZpndc ___ 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 Restart after crash
There is a bug with safe_asterisk and FC2, you must edit the script to remove 'daemon' from the the startup command and then it will auto restart. Craig - Original Message - From: David Phelan [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, April 22, 2005 1:32 PM Subject: RE: [Asterisk-Users] Asterisk Restart after crash After a crash of what?? Linux...asterisk?? Depends on how you have it setup If you start asterisk with safe_asterisk, then if asterisk crashes it will start again. If you run safe_asterisk from say...your rc.local then it will start when linux restarts. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Smith Sent: Friday, 22 April 2005 1:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk Restart after crash Does Asterisk restart itself if it crashes? If not is there a way to make linux do it? ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 21/04/2005 ___ 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] Line Noise UPDATE - If you've got line noise, read this
I have three in one machine, and 4 customers that have 2 in each of their machines. The only problem I've ever had is momentary echo when a call first begins, but that is to be expected until the line trains. - Original Message - From: Paul [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, April 22, 2005 2:40 AM Subject: RE: [Asterisk-Users] Line Noise UPDATE - If you've got line noise,read this Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using X100P cards and NOT having this problem?? Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Sent: Thursday, April 21, 2005 14:23 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Line Noise UPDATE - If you've got line noise,read this Hm...well, here's something interesting. On my previous box, neither of the cards were sharing IRQs with anything.now, both cards are on 11, along with many other things. This could very well be a problem. As far as the network goes, there is very little traffic and the switch is full duplex 100 megabit. Bandwidth is only a factor on he local lan, since asterisk dials out through a FXO card(X100P) then it doesn't go across my broadband connection. Let me know your thoughts.sorry for the verbose output. Paul [EMAIL PROTECTED] /sbin]$ ./lspci -v 00:00.0 Host bridge: VIA Technologies, Inc. VT8377 [KT400 AGP] Host Bridge (rev 80) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master, 66Mhz, medium devsel, latency 8 Memory at d000 (32-bit, prefetchable) [size=128M] Capabilities: available only to root 00:01.0 PCI bridge: VIA Technologies, Inc.: Unknown device b198 (prog-if 00 [Normal decode]) Flags: bus master, 66Mhz, medium devsel, latency 0 Bus: primary=00, secondary=01, subordinate=01, sec-latency=0 I/O behind bridge: 9000-9fff Memory behind bridge: e800-e9ff Prefetchable memory behind bridge: d800-e7ff Capabilities: available only to root 00:09.0 Communication controller: Tiger Jet Network Inc. Intel 537 Subsystem: Intel Corp.: Unknown device 0003 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at a000 [size=256] Memory at eb002000 (32-bit, non-prefetchable) [size=4K] Capabilities: available only to root 00:0a.0 Communication controller: Tiger Jet Network Inc. Intel 537 Subsystem: Intel Corp.: Unknown device 0003 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at a400 [size=256] Memory at eb00 (32-bit, non-prefetchable) [size=4K] Capabilities: available only to root 00:0b.0 Ethernet controller: Linksys Network Everywhere Fast Ethernet 10/100 model NC100 (rev 11) Subsystem: Linksys: Unknown device 0570 Flags: bus master, medium devsel, latency 32, IRQ 5 I/O ports at a800 [size=256] Memory at eb001000 (32-bit, non-prefetchable) [size=1K] Expansion ROM at unassigned [disabled] [size=128K] Capabilities: available only to root 00:0f.0 RAID bus controller: VIA Technologies, Inc.: Unknown device 3149 (rev 80) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master, medium devsel, latency 32, IRQ 11 I/O ports at ac00 [size=8] I/O ports at b000 [size=4] I/O ports at b400 [size=8] I/O ports at b800 [size=4] I/O ports at bc00 [size=16] I/O ports at c000 [size=256] Capabilities: available only to root 00:0f.1 IDE interface: VIA Technologies, Inc. VT82C586/B/686A/B PIPC Bus Master IDE (rev 06) (prog-if 8a [Master SecP PriP]) Subsystem: VIA Technologies, Inc. VT8235 Bus Master ATA133/100/66/33 IDE Flags: bus master, medium devsel, latency 32 I/O ports at c400 [size=16] Capabilities: available only to root 00:10.0 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00 [UHCI]) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at c800 [size=32] Capabilities: available only to root 00:10.1 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00 [UHCI]) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master, medium devsel, latency 32, IRQ 10 I/O ports at cc00 [size=32] Capabilities: available only to root 00:10.2 USB Controller: VIA Technologies, Inc. USB (rev 81) (prog-if 00 [UHCI]) Subsystem: Elitegroup Computer Systems: Unknown device 1884 Flags: bus master,
Re: [Asterisk-Users] Re: Email to Fax
You could try http://www.inter7.com/?page=astfax - I haven't used it yet myself but it looks like it'll work. Craig - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Justin Newman' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, April 22, 2005 2:13 PM Subject: RE: [Asterisk-Users] Re: Email to Fax How are you doing it Justin? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Newman Sent: Jueves, 21 de Abril de 2005 10:53 p.m. To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: Email to Fax Message: 11 Date: Thu, 21 Apr 2005 20:39:22 -0500 From: Anton Krall [EMAIL PROTECTED] Subject: [Asterisk-Users] Email to Fax Anybody doing email to fax using spandsp? Yep... Justin Newman Newman Telecom, Inc. [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 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] Error in starting asterisk
channel=1-15,17-30 Apr 22 06:02:04 WARNING[18915]: parse error: No category context for line 10 of zapata.conf Apr 22 06:02:04 ERROR[18915]: Unable to load config zapata.conf Just a couple of guesses here... I'm not so sure the line 10 is counting correctly (or you're not counting the blank lines) as the echotraining=800 is just fine/valid. Check the following items - do you have a from-pstn context in your extensions.conf? - change your channel statement to channel=1 and see if it loads (just to help ID which statement is causing the problem) - same for signalling, change it to another valid value for testing - remove all the unneeded statements (for testing only), including faxdetect, usecallerid, echo statements, group. All of those items will default to something acceptable for startup purposes. Run ztcfg -vv and post the results. Might also consider starting asterisk from the linux command line with asterisk -c and review the many messages looking for exceptions. If none of the above helps ID the problem, copy/paste the exact 20 or so lines from zapata.conf (don't delete anything) and repost. ___ 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] Line Noise UPDATE - If you've got line noise, read this
Paul wrote: Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using X100P cards and NOT having this problem?? Yes. I use X100P in at least several different Asterisk system and have no problems. One of them is an Intel motherboard, one of them is a Supermicro, one of them is ASUS motherboard, one of them is an older Compaq system. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 Restart after crash
Could you give some more information on where to remove 'daemon' and the effects? Since all our productionservers running FC2 I'm a bit concerned. There is a bug with safe_asterisk and FC2, you must edit the script to remove 'daemon' from the the startup command and then it will auto restart. Thanks a lot Guido Hecken ___ 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 Restart after crash
OK how do you know if it's running in safe_asterisk mode? I am running [EMAIL PROTECTED] Does that run in safe mode by default? What file do you look at to see how asterisk starts up? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Phelan Sent: Friday, April 22, 2005 1:32 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk Restart after crash After a crash of what?? Linux...asterisk?? Depends on how you have it setup If you start asterisk with safe_asterisk, then if asterisk crashes it will start again. If you run safe_asterisk from say...your rc.local then it will start when linux restarts. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Smith Sent: Friday, 22 April 2005 1:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Asterisk Restart after crash Does Asterisk restart itself if it crashes? If not is there a way to make linux do it? ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 21/04/2005 ___ 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] IAX2 Error
On April 22, 2005 08:33 am, Robson Ribeiro wrote: Apr 22 11:50:19 WARNING[9124]: Received mini frame before first full voice frame It means nothing, only that the first two packets out of the gate arrived in the opposite order that they were sent. Simple explanation: Asterisk sends small (mini) frames whenever possible. This is most of the time. But in order to send mini frames, it must send the occassional large (full) frame in order to establish a reference point for the mini ones. At the start of a call, Asterisk sends a full frame, then starts sending mini frames. All this message means is that the full frame didn't arrive first. That's it. There is nothing you can tweak to fix it; it's just the nature of the beast, which is why it's a warning and not an error. -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] Asterisk transcoding
I would like to learn more over the transcoding function with asterisk. How exactly works asterisk, in order to transcoding. Where I can get exactly informations? If asterisk transcodes, for example ilbc to gsm, as I can see which (ilbc) rtp-packet becomes which (gsm) rtp-packet? would be very grateful for assistance -- +++ GMX - Die erste Adresse für Mail, Message, More +++ 1 GB Mailbox bereits in GMX FreeMail http://www.gmx.net/de/go/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
Re: [Asterisk-Users] Line Noise UPDATE - If you've got line noise, read this
On April 22, 2005 03:40 am, Paul wrote: Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using X100P cards and NOT having this problem?? Have you tried my suggestion regarding using only ONE X100P to test? Or about trying a TDM02P (and seeing if Digium can give you a 30 day trial)? Or about calling Digium for support since these are actual official Digium X100P cards and not some cheap knockoff you got on ebay, especially since this list is littered with warnings about them and you have likely spent 10x the cost of an official X100P from Digium if you factor in your time and aggravation? Not trying to be an ass, and if I'd missed your messages about having already tried these venues I apologize. -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] callto: URL (URI) tag for dialing
I see that there seems to be a 'callto' URL/URI for dialling a phone number... ie - on my web site's Contact Page - I have added the code... a href=callto:+27128070590+27 12 807-0590/a There should be some generic way for Mozilla (firefox - etc) to somehow turn a click on such a link into persuading Asterisk to dial the number for me and connect it to my SIP hard-phone. 1 - mini application under mozilla to collect the number/sip address, add in a static local extension (personal settings?) and pass info to a listener (auto-dialer) on the Asterisk Machine 2 - Auto Dialer dials my extension, then on answer, dials the URL or phone number. The URL could either be a simple phone number or a full SIP address?? Anyone done this? ..and care to share? -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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: chan_unicall.c compile error
Fabio, check if you have libtiff and libtiff-devel installed. and also you have to patch the asterisk source code first.ç I dont know if you did that... regards, Hector. On 4/21/05, Fabio Vasco [EMAIL PROTECTED] wrote: Hector, This is my Linux Fedora Core 3 version info [EMAIL PROTECTED] proc]# cat version Linux version 2.6.9-1.667 ([EMAIL PROTECTED]) (gcc version 3.4.2 20041017 (Red Hat 3.4.2-6.fc3)) #1 Tue Nov 2 14:41:25 EST 2004 I am get Asterisk with the cvs -r stable, i supose the version is 1.0.6. Thanks for your help. Fabio _ MSN Busca: fácil, rápido, direto ao ponto. http://search.msn.com.br ___ 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] Mysql using Sip and voicemail
I am currently running asterisk 1.0.7 and decided to try using MySQL to hold some of my voicemail and sip configuration. As a note - MySQL is already holding my CDR info. I followed the directions in on voip-info.org to copy files, modify Makefiles, recompile, and change the conf files accordingly. I have run into a few bumps that I need to ask about. With the voicemail database, if the voicemailbox is in the default context, all is well. If I attempt to place the mailbox in any other context, it will not work at all like the mailbox does not exist. The other issue is with the sip database, which does not to appear to work at all. When attempting to connect a SIP phone that setup in the database, I receive the error Registration from '999 sip:[EMAIL PROTECTED]' failed for '192.168.0.129' When starting asterisk, I can see where asterisk is logging into the MySQL database correctly. I have double checked the configuration files, database structure, and even tried setting the context as default (since that worked for voicemail) Any suggestions on what to check into Thanks Ben Johnson ___ 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] using * for Internet call waiting
The * box would sit in a CO connected via PRIs. Gary Gary Carr wrote: Wondering if it is possible or if something already exist to setup * to offer Internet Call Waiting. For those that do not know what it is, it's a small application that runs on a users computer that will pop up a window letting them know they have a incoming call and who it is from then they can choose to take the call which will disconnect their dialup modem and ring their phone or send the call to voice mail. That doesn't really make sense if the * box is in your house because if the phone line is tied up for a dialup call, then the * box doesn't have a phone line to receive the call either (unless you had call hunting in which case you wouldn't need the feature in the first place). This sounds like the sort of feature that can only be offered on the central office side which can know your line is tied up and then know to email/alert you. The other scenario is having an * box in a call center that is forwarding calls to agents and notifies them by TCP/IP if when it tries their extension and gets a busy signal. This sounds possible, but I don't think it's what you meant. 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 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] ASTCC
Chris, There is no official documentation, but here is what I have found in the control panel. *BRANDS* This is where you can setup different cards with different Service fees. I'm not sure what the INC column is for, I usually leave it set at 6. I think it sets 6 seconds to the minimum bill time. Service fees and Servie Fee Days is for like a monthly charge. I havent figured out where the Markup field comes into play yet. *CARDS* This is where you make the cards. You can get a list of cards or you can make/add money to any card. You are able to use GET style URLs to make your own interface to this. Just add a card and notice the URL. This could be very helpful if you want to build another interface (I just made direct DB calls) *TRUNKS* This is where you setup your Trunks(duh?). You can name the trunk, set it technology, and then relate it to a real trunk/peer name on your asterisk box. If I understand correctly, only SIP and IAX work, but I could be wrong. I know SIP works, I have used it. *ROUTES* This is a decent attempt at a LCR script. Here you will setup your costs for different providers. For example, if you want calls to Mexico to me $.10 a minute, then in the Pattern field, put '^01152.*' (without the '') And in the Cost per additional minute, put 1000. (The costs are in 1/100th of a penny. There is so much that can be done with the Routes. You can specify more than one trunk so if it is possible to go out a cheap provider for one area, then put that first, if not, then it will try the next one. You are also able to charge a Connect Fee and Include X amount of seconds with that fee. *CONFIGURE* This is where you setup your DB connection and some other information. Host, Username, and Password are all related to your DB. Card length is how many digits the card number will be. I think the voiceover always says 12 digits (Not sure). The Email New Card Info did not work for me and I left everything else set to no because I didnt neet it. *IAX and SIP FRIENDS* I'm sorry, but I dont know what those are for, I havent found a need for them, but maybe my setup doesnt need them. *CDRs* Out-of-the-box CDRs do not work. They are broken. Unless another version was released since I d/led it. I just updated the ASTCC entry on the voip-info.org wiki and the quick-fix is there. http://voip-info.org/tiki-index.php?page=ASTCC *PROBLEMS* Besides the CDRs, the only problems I have found with ASTCC is that at the 'one minute warning' ASTCC cuts into the call, announces that you have one minute left, then the call is supposed resume for your last minute. If I had to guess, the RTP stream is broken when this happens because after the warning, neither side can hear the other. Unfortunately, I havent had a chance to find out why the voice traffic stops. If anyone could let me know what they find I would appreciate it. I asked Digium about it, and they wanted to charge me their hourly rate to work on it. *OTHER NOTES* If you look on the voip-info.org, there are a couple neat ideas to use with AstCC. I have a box setup that when you dial out, it asks for the pin number then if the pin is right, it will go thru, if not, it denies you. There is also a way to make it go off of the caller ID so no pin is needed. These are just some things that I have found working with AstCC. I am not an expert by any means, but if you understand Perl, then the AGI script should be fairly easy to modify to suit your needs. I hope this 'guide' is helpful to someone. I would like to hear of other people's experience with ASTCC. Dave Kettmann NetLogic 314-266-4000 -Original Message- From: Chris [mailto:[EMAIL PROTECTED] Sent: Thursday, April 21, 2005 1:48 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ASTCC Is there any documentation on how to setup the ASTCC?I've got it working, but I don't quite understand what the web interface is referring to. Chris ___ 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] using * for Internet call waiting
The TDM part is pretty simple. The end user needs the call forward busy feature on thier line with the calls being forwarded to the * server. Taking it from there and sending it to a app on the users machine is whats left. I was thinking it could be sent with sip and a long timeout value. Gary I've seen this service done with AOL, I was curious how it was done on standard phone lines. Was it something the coordinated with the telco in some sort of hunt group configuration or something of that nature? - Dan Levine CYTEXONE | Your Technology Specialists t: 877.CYTEXONE x 810 l: 212.477.0990 x 810 e: [EMAIL PROTECTED] http://www.cytexone.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr Sent: Thursday, April 21, 2005 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] using * for Internet call waiting Wondering if it is possible or if something already exist to setup * to offer Internet Call Waiting. For those that do not know what it is, it's a small application that runs on a users computer that will pop up a window letting them know they have a incoming call and who it is from then they can choose to take the call which will disconnect their dialup modem and ring their phone or send the call to voice mail. Thanks, Gary ___ 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] using * for Internet call waiting
That's pretty close to what we are looking for but we want the user to have
the option of taking the call which would disconnect the modem connection
and allow the call to ring thru to the phone. Not sure how to accomplish
that. I am sure our programmer could code a client but he has no experience
with *. If we can figure out that part we could come up with something.
Gary
I'm an ISP, what I would like is a client for the dialup customer to run.
They would use call fwd busy to my did on an asterisk box.
I'd signal and they could click on button (URL) to download .wav file in
asterisk voice mail.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mojo with
Horan Company, LLC
Sent: Thursday, April 21, 2005 5:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] using * for Internet call waiting
I once tried the pagoo service. Seems I had to ask the telco for Call
Forward Busy, and provide them with the toll free number pagoo gave me
for their service. When the forwarded call is received by their
systems, they would see _my_ callerid information, and thus know to
contact my computer for the notification purpose.
Also, not sure if this is on track with what you want, but I've used
jabber_client.pl tied into my dialplan to popup the callerid info of an
incoming call on my screen.. I could then choose to answer the call or
let it ring to voicemail. Seems the jabber client Neos has
well-designed popups.
links:
http://jabberd.jabberstudio.org/2/
for the jabber_alert.pl script, allows sending jabber msgs from cmd line.
http://www.neosmt.com/
for a jabber client that pops up incoming messages. Note, this is also
an H.323 client. Haven't tried it with * yet, but I have been meaning to.
Here's the specific Dialplan line I use:
[inpstn]
exten = s,2,TrySystem(echo Incoming call from :${CALLERID} |
jabber_alert.pl -e [EMAIL PROTECTED] -n [EMAIL PROTECTED] -w
senders_password)
Because it can sometimes take 2 or 3 seconds to send the jabber message
on my network, I use TrySystem instead of System, which blocks, waiting
for the return code from the command I passed. Because the return code
is prolly irrelevant, you'd most likely want to use TrySystem too...
hope this helps :)
Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr
Sent: Thursday, April 21, 2005 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] using * for Internet call waiting
Wondering if it is possible or if something already exist to setup * to
offer Internet Call Waiting. For those that do not know what it is, it's
a small application that runs on a users computer that will pop up a
window letting them know they have a incoming call and who it is from
then they can choose to take the call which will disconnect their dialup
modem and ring their phone or send the call to voice mail.
Thanks,
Gary
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RE: [Asterisk-Users] ASTCC
For everyone's information and so it is on the list somewhere, there is a copy of this at http://voip-info.org/tiki-index.php?page=ASTCCGuide This also includes the explanation of the CDR problem. -Original Message- From: Dave Kettmann Sent: Friday, April 22, 2005 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] ASTCC Chris, There is no official documentation, but here is what I have found in the control panel. *BRANDS* This is where you can setup different cards with different Service fees. I'm not sure what the INC column is for, I usually leave it set at 6. I think it sets 6 seconds to the minimum bill time. Service fees and Servie Fee Days is for like a monthly charge. I havent figured out where the Markup field comes into play yet. *CARDS* This is where you make the cards. You can get a list of cards or you can make/add money to any card. You are able to use GET style URLs to make your own interface to this. Just add a card and notice the URL. This could be very helpful if you want to build another interface (I just made direct DB calls) *TRUNKS* This is where you setup your Trunks(duh?). You can name the trunk, set it technology, and then relate it to a real trunk/peer name on your asterisk box. If I understand correctly, only SIP and IAX work, but I could be wrong. I know SIP works, I have used it. *ROUTES* This is a decent attempt at a LCR script. Here you will setup your costs for different providers. For example, if you want calls to Mexico to me $.10 a minute, then in the Pattern field, put '^01152.*' (without the '') And in the Cost per additional minute, put 1000. (The costs are in 1/100th of a penny. There is so much that can be done with the Routes. You can specify more than one trunk so if it is possible to go out a cheap provider for one area, then put that first, if not, then it will try the next one. You are also able to charge a Connect Fee and Include X amount of seconds with that fee. *CONFIGURE* This is where you setup your DB connection and some other information. Host, Username, and Password are all related to your DB. Card length is how many digits the card number will be. I think the voiceover always says 12 digits (Not sure). The Email New Card Info did not work for me and I left everything else set to no because I didnt neet it. *IAX and SIP FRIENDS* I'm sorry, but I dont know what those are for, I havent found a need for them, but maybe my setup doesnt need them. *CDRs* Out-of-the-box CDRs do not work. They are broken. Unless another version was released since I d/led it. I just updated the ASTCC entry on the voip-info.org wiki and the quick-fix is there. http://voip-info.org/tiki-index.php?page=ASTCC *PROBLEMS* Besides the CDRs, the only problems I have found with ASTCC is that at the 'one minute warning' ASTCC cuts into the call, announces that you have one minute left, then the call is supposed resume for your last minute. If I had to guess, the RTP stream is broken when this happens because after the warning, neither side can hear the other. Unfortunately, I havent had a chance to find out why the voice traffic stops. If anyone could let me know what they find I would appreciate it. I asked Digium about it, and they wanted to charge me their hourly rate to work on it. *OTHER NOTES* If you look on the voip-info.org, there are a couple neat ideas to use with AstCC. I have a box setup that when you dial out, it asks for the pin number then if the pin is right, it will go thru, if not, it denies you. There is also a way to make it go off of the caller ID so no pin is needed. These are just some things that I have found working with AstCC. I am not an expert by any means, but if you understand Perl, then the AGI script should be fairly easy to modify to suit your needs. I hope this 'guide' is helpful to someone. I would like to hear of other people's experience with ASTCC. Dave Kettmann NetLogic 314-266-4000 -Original Message- From: Chris [mailto:[EMAIL PROTECTED] Sent: Thursday, April 21, 2005 1:48 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ASTCC Is there any documentation on how to setup the ASTCC?I've got it working, but I don't quite understand what the web interface is referring to. Chris ___ 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
[Asterisk-Users] Asterisk acting as PBX + SIP Proxy ... possible?
Hello, I'm in the process of implementing the following setup External SIP phones at another location(s) (nat = yes) | | Analog phone line | | |-- |ext if 142.x.x.41 | |Asterisk | |int if 192.168.0.1 |-- | Internal SIP Phones (nat=no) Excuse my ASCII art ... if you cant see the diagram I'm basically doing the following: - There are some phones on the LAN, and some other phones on the internet side - Both sets of phones use Asterisk to make calls between each other as if they were all on LAN and to the phone line. Is something like this going to work reliably? Or will I need a second central server to act as a proxy. The reason I'm asking this is that I have been able to make this setup work but am having some strange registration issues whenever my external sip phones sit behind Linksys router (I get 403 forbidden) ... when I use some other router the stuff seems to work. But I'm worried about reliability since I read recently that Asterisk is not a proxy and I'm definitely using it as an outgoing proxy in this case. http://www.voip-info.org/wiki-Asterisk+SIP+not-proxy Has anyone successfully created this kind of setup before? (having Asterisk pass calls on both LAN and WAN side?) Do you have any hints for me to get this 403 forbidden error figured out? I think it might have something to do with FQDN - but the strange thing is that it happens only behind Linksys And if I do need an outgoing proxy which proxy do you recommend? Thank you, Tomas ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] One touch voicemail on Cisco 7940/60
On Fri, 2005-04-22 at 11:13 +0100, Simon Morris wrote:
On Thu, 2005-04-21 at 21:36 +0100, Ron Wellsted wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Morris, Simon wrote:
Hello,
I'd like to program my Cisco phones to authenticate themselves to
voicemail upon hitting the right button on my 7940/60's
Ideally the voicemail app will detect which extension the call is
coming
from and drop the user straight into the menu.
Is this possible?
Many thanks
~sm
Yes this is possible.
In your extensions.conf:
exten = _8501,1,Answer()
exten = _8501,2,VoicemailMain(s${CALLERIDNUM})
exten = _8501,3,Hangup()
then program the messages button to dial 8501 either via settings,
SIP
Settings, Messages URI (set to 8501) or in the SIPDefault.cnf file
Guys - sorry to change my original question.
That solution does exactly what I asked BUT! I'd like it to dial and
know which extension I'm coming from and then prompt for the password.
Just realised that the solution above allows people to wander around
listening to other peoples voicemail at the press of the button :-)
So.. how to bypass the Enter your mailbox number stage in voicemail
and go straight to the password prompt.
Thanks guys
~sm
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[Asterisk-Users] No such context/extension
To All,
I am a new to Asterisk and dialplans have me stumped I just inherited 2
Asterisk servers conected as IAX peers.
Now from what i can tell when Asterisk Server (ask-CHIC) needs to make a
call to an extension which resides on the other server (ask-MAIN) it
goes over a IAX channel.
Now i am trying to add that third asterisk server to mix (ask-SD) and i
figured i would do it in baby steps.
The first thing i did was configure two local SIP client so they could
call each other and leave voicemail and that works just fine.
I then tried to add Asterisk-Server(ask-SD) to (ask-Main) as a IAX
peer. Just like (ask-chic).
To test i tried dialing an (ask-SD) ext from a phone off the (ask-MAIN)
server.
The call did not go through. However i watched it from the CLI and
captured the following output. What could be wrong. i am so stumped.
'[EMAIL PROTECTED]' in 15000 ms
-- Accepting call from '' to '7101' on channel 0/23, span 2
-- Executing Dial(Zap/47-1, IAX2/ask-SD/7101) in new stack
-- Called ask-SD/7101
Apr 21 13:52:12 WARNING[147465]: chan_iax2.c:5495 socket_read: Call
rejected by ask-SD: No such context/extension
-- IAX2/ask-SD/2 is circuit-busy
-- Hungup 'IAX2/ask-SD/2'
== Everyone is busy/congested at this time
-- Executing Congestion(Zap/47-1, ) in new stac
here is server (ask-sd) iax.conf and extension.conf files
iax.conf:
[EMAIL PROTECTED] asterisk]# cat iax.conf
[general]
allow=all
jitterbuffer=no
tos=lowdelay
[guest]
type=user
context=guest
callerid=Gust User
; BMS-ask-Main-asterisk - Incoming -
;
[ask-mail]
type=user
secret=ask-mail
context=from-ask-main
disallow=all
allow=ulaw
; bmc-asl-main - Outgoing
;
[telx-nyc]
type=peer
username=ask-sd ; our username
secret=ask-sd; our password
host=192.168.11.30 ; host to connect to
;qualify=yes
;trunk=yes ; use trunking
[EMAIL PROTECTED] asterisk]#
Extension.conf
[EMAIL PROTECTED] asterisk]# cat extensions.conf
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
exten = 7101,1,Dial(SIP/7101,20)
exten = 7101,2,Voicemail(u7101)
exten = 7101,102,Voicemail(b7101)
exten = 7101,103,Hangup
exten = 7102,1,Dial(SIP/7102,20)
exten = 7102,2,Voicemail(u7102)
exten = 7102,102,Voicemail(b7102)
exten = 7102,103,Hangup
exten = 7199,1,VoicemailMain(${CALLERIDNUM})
[macro-telx-nyc]
exten = s,1,Noop()
exten = s,2,Dial(IAX2/ask-mail/${ARG1})
[outgoing]
;ingnorepat = 9
exten = _9NXXNXX,1,Noop()
exten = _9NXXNXX,2,Macro(ask-main,${EXTEN})
exten = _9NXXNXX,3,Playback(invalid)
exten = _9NXXNXX.4,Hangup
[EMAIL PROTECTED] asterisk]#
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[Asterisk-Users] Alcaterl IP-touch phones
Hi all, Has any one tested Asterisk with the new Alcatel IP-touch phones (IP phones with xml) Thanks ___ 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] No such context/extension
On 22/04/05, MDM [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] asterisk]# cat iax.conf ; BMS-ask-Main-asterisk - Incoming - ; [ask-mail] type=user secret=ask-mail context=from-ask-main disallow=all allow=ulaw There are probably some typos in there which might be adding to your problems - mail vs main - but the bigger problem is that you are sending calls to a context called 'from-ask-main', but that context doesn't exist in your extensions.conf. You have one called 'from-sip' which is where you probably could send them. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[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] Debugging zaphfc + PBX integration
Gavin Hamill wrote: I know the cables themselves are wired correctly because our local PBX support made them, and they work perfectly when plugged into a real BT ISDN2e wallbox it seems as if this is exactly your problem. the wallbox has a NT pinout = straight trough cable the hfc card has a TE pinout = you need a cross-over (isdn not ethernet!!) cable to connect to your local pbx which also has a TE pinout. the nt/te switches on the hfc card do not cross over the rx/tx pairs of the card. this has to be done with the cabling. i don't think you will need any termination resistors if your cable is only a few meters and does not have any other devices on the bus. regards frank sautter ___ 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] Dynamic queue member behaviour
I found that if I dynamically add, for example SIP/8000, to a queue, then calls in the queue will sorta pile up on the 9 extensions on that phone - not what we want to happen. If I log in to the queue using AgentLogin, then the behaviour is as expected - one call at a time. Is there a way around this, or am I adding dynamically to the queue incorrectly? Thanks in advance for any assistance. ___ 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] One touch voicemail on Cisco 7940/60
On Fri, April 22, 2005 9:27 am, Simon Morris said: So.. how to bypass the Enter your mailbox number stage in voicemail and go straight to the password prompt. Remove the s at the beginning of the argument to VoiceMailMain() http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20VoiceMailMain Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ 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] One touch voicemail on Cisco 7940/60
That solution does exactly what I asked BUT! I'd like it to dial and
know which extension I'm coming from and then prompt for the password
exten = _8501,2,VoicemailMain(s${CALLERIDNUM})
Just remove the 's' from the line above. Not to sound like a smart ass, but
this is all very well documented in the wiki. Please check there before
posting to the list it would save a lot of people a lot of time trying to
show you how to do this. I don't mind helping but please try to help
yourself first.
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Re: [Asterisk-Users] Alcaterl IP-touch phones
I don't believe they speak SIP - not yet, that is ;) Herve. - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, April 22, 2005 3:36 PM Subject: [Asterisk-Users] Alcaterl IP-touch phones Hi all, Has any one tested Asterisk with the new Alcatel IP-touch phones (IP phones with xml) Thanks ___ 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] Asterisk Restart after crash
In your asterisk script in init.d that calls safe_asterisk change this:
start() {
# Start daemons.
echo -n $Starting asterisk:
if [ -f $SAFE_ASTERISK ] ; then
DAEMON=$SAFE_ASTERISK
fi
if [ $AST_USER ] ; then
ASTARGS=-U $AST_USER
fi
if [ $AST_GROUP ] ; then
ASTARGS=`echo $ASTARGS` -G $AST_GROUP
fi
--daemon $DAEMON $ASTARGS
RETVAL=$?
[ $RETVAL -eq 0 ] touch /var/lock/subsys/asterisk
echo
return $RETVAL
}
to this:
start() {
# Start daemons.
echo -n $Starting asterisk:
if [ -f $SAFE_ASTERISK ] ; then
DAEMON=$SAFE_ASTERISK
fi
if [ $AST_USER ] ; then
ASTARGS=-U $AST_USER
fi
if [ $AST_GROUP ] ; then
ASTARGS=`echo $ASTARGS` -G $AST_GROUP
fi
$DAEMON $ASTARGS
RETVAL=$?
[ $RETVAL -eq 0 ] touch /var/lock/subsys/asterisk
echo
return $RETVAL
}
ie remove 'daemon' from the command.
Test it by kill -9 asterisk pid and see if it restarts - it is quite
aggressive.
Craig
- Original Message -
From: Guido Hecken [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, April 22, 2005 7:56 PM
Subject: RE: [Asterisk-Users] Asterisk Restart after crash
Could you give some more information on where to remove 'daemon' and the
effects?
Since all our productionservers running FC2 I'm a bit concerned.
There is a bug with safe_asterisk and FC2, you must edit the script to
remove 'daemon' from the the startup command and then it will auto
restart.
Thanks a lot
Guido Hecken
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[Asterisk-Users] can't make my PRI dial out
I have a full PRI installed on my * machine. I can get inbound calls
just fine but can't make outbound ones.
Zaptel.conf says;
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
zapata.conf says
language=en
context=default
switchtype=4ess
pridialplan=unknown
signalling=pri_cpe
channel=1-23
echocancel=yes
group=1
dial string in extensions.conf says
; calls to the outside world via the PSTN
exten = _81NXXNXX,1,Dial(ZAP/1/${EXTEN:1})
When I try to dial a number I get
- Executing Dial(SIP/3710-23ea, ZAP/17327356701) in new stack
Apr 22 10:19:17 NOTICE[28197]: app_dial.c:803 dial_exec: Unable to
create channel of type 'ZAP' (cause 0)
== Everyone is busy/congested at this time
pri show span 1 says
120b-pbx*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Up, Active
Switchtype: ATT 4ESS
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3
Paid support from Digium sucks royally. They don't even know what their
own error codes mean!!
Any ideas?
Thanks
Mark
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Re: [Asterisk-Users] DTFM tones almost completly muted.
On 22/04/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Fri, 22 Apr 2005, Peter Bowyer wrote: On 22/04/05, Ian Hailey [EMAIL PROTECTED] wrote: Hello everyone, I am trying to receive DTMF commands on asterisk from PSTN calls terminated at my asterisk box. I have tried to terminate the PSTN calls with both SIP and IAX using sigate.co.uk and voipuser as the PSTN terminator. When I listen to tones sent from the PSTN side (e.g. continuous DTMF tone of about 3 seconds) on the asterisk server (stored in the voice mail) the tone is more or less completely muted, just the initial tone start can be heard. I am using the G711 codec. Does anyone have any idea if these tones are on purpose muted by the service providers or any other reason why it does not work? Most likely the DTMF tones have been detected at the point where the call was converted PSTN-SIP/IAX, and forwarded instead as an indication (ie via SIP INFO or RFC2833 or whatever. So you won't hear them in a recording of the audio stream. The remaining blip is just the little bit at the start before the gateway recognised the tone. You should receive the indication in your SIP or IAX connection and Asterisk should see it (but its not audio any more). No, it doesn't work, period. Somehow Sipgate eats the DTMF. No amount of messing with the DTMFMode settings in Asterisk helps. Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[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] can't make my PRI dial out
On April 22, 2005 11:48 am, Mark Phillips wrote: Nothing happens. I get the same (non)error. I get plenty of output when receiving a call however. Odd... Here is my zapata.conf setup for my PRI: --- [channels] context=BellPRI switchtype=national pridialplan=unknown priindication=outofband overlapdial=no signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=no relaxdtmf=no group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived channel = 1-23 --- I simply Dial(Zap/g1/5551212). The fact that it doesn't think it can pick up a channel is interesting, I haven't run across that before. Combined with the fact that you can receive calls just fine, this is a very strange little problem. With my config (modified for your switchtype and context ONLY), what do you get when you try to dial a number? -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
RE: [Asterisk-Users] QOS Routers
You may want to check out edgewaternetworks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Friday, April 22, 2005 11:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] QOS Routers Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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 ___ 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] Rejected connect attempt
Hello, I have seen the following in my log files. For the life of me I can not work out why. Apr 22 22:10:40 NOTICE[19236] chan_iax2.c: Rejected connect attempt from 65.39.205.121, who was trying to reach 'i@' Would someone explain why, or point me in the direction I can read about it? Many thanks, Spencer --- https://www.dalmany.co.uk/dundi/dundi.php https://www.dalmany.co.uk/asterisk/index.php ___ 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] Mysql using Sip and voicemail
Actually, I just got the SIP phones working right before I received. I am not exactly sure what I changed, but I basically went through and recreated, recompiles, etc then it worked. I will keep the sip debug in mind if I have any more sip problems. Thanks for your help. Still can not figure out the voicemail context problem though From: Race Vanderdecken [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Mysql using Sip and voicemail Date: Fri, 22 Apr 2005 12:17:38 -0400 I can give you some help with the SIP stuff. Try it again with sip debug turned on and send the output back here. It would be good to see the SIP messages that are being transferred. The Asterisk SIP Stack is good, but not great. You might need to just add or delete an option in the sip.conf file. I did a very similar thing with SIP config and app_radius stuff. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Johnson Sent: Friday, April 22, 2005 8:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Mysql using Sip and voicemail I am currently running asterisk 1.0.7 and decided to try using MySQL to hold some of my voicemail and sip configuration. As a note - MySQL is already holding my CDR info. I followed the directions in on voip-info.org to copy files, modify Makefiles, recompile, and change the conf files accordingly. I have run into a few bumps that I need to ask about. With the voicemail database, if the voicemailbox is in the default context, all is well. If I attempt to place the mailbox in any other context, it will not work at all like the mailbox does not exist. The other issue is with the sip database, which does not to appear to work at all. When attempting to connect a SIP phone that setup in the database, I receive the error Registration from '999 sip:[EMAIL PROTECTED]' failed for '192.168.0.129' When starting asterisk, I can see where asterisk is logging into the MySQL database correctly. I have double checked the configuration files, database structure, and even tried setting the context as default (since that worked for voicemail) Any suggestions on what to check into Thanks Ben Johnson ___ 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] Debugging zaphfc + PBX integration
On Friday 22 April 2005 14:46, Frank Sautter wrote: Gavin Hamill wrote: it seems as if this is exactly your problem. Sorry Frank, but this one isn't as simple as cabling... I've made reference in this thread already that I do have both straight + ISDN crossover (3/4 and 5/6 swapped) cables, and none of them work... one will get further than the other, i.e. using 'dmesg' I see a TEI request *FROM* the PBX, but I don't see any output from the HFC card going back to the PBX to tell it what TEI to use... Cheers, Gavin. ___ 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] Line Noise UPDATE - If you've got line noise, read this
We have basically the same setup. My cards are on 5 and 7 as well and I've disabled EVERYTHING is the bios that is not necessary; USB, serial, parallel, ect. I would think that if it was an IRQ issue, the call wouldn't tank when I connected it on the card with it's own IRQ. I just got in my new Cisco 7940 a few minutes ago and when I get a powersupply for it, I'm going to remove the Sipura-841 and try this one out and see if maybe that doesn't fix the problem. I'm doubtful, but it seems that this POS sipura is the only thing that is REALLY differing from out configurations. I didn't install X when I setup this box, so I know it's not running, but it's good to know that it can cause problems. Thanks for the info, every bit helps. Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed Sent: Friday, April 22, 2005 05:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Noise UPDATE - If you've got line noise,read this On Fri, Apr 22, 2005 at 02:40:10AM -0500, Paul said: Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using X100P cards and NOT having this problem?? Yes. I can make a call from a POTS phone hooked up to a Cisco ATA 186, out one X100P to the PSTN, back in a second X100P, to a phone hooked up to the second port on the ATA186 with no noise, and no echo, and a pretty small delay (which you can hear with one handset in each ear.) I have disabled most of the on-board I/O such as parallel, serial, and extra USB controllers, and the X100's are on int 5 and 7, not shared with anything. Interrupts 10 and 11 have a bunch of stuff shared and are used by USB controllers, ethernet ports (one on each IRQ) video card, SCSI controller, and one unknown device (some special nVidia device.) This machine is also used as a firewall / gateway / email server but does NOT run X (which I hear can cause problems on some machines.) I've been running this configuration for about 9 months with virtually no problems in a SOHO environment including weekly 3-hour long conference calls. I realize this doesn't help you much, but it IS possible for the configuration to work. I have been thinking about getting a Sipura 3000 to add another FXS port and remove one X100P which would also cut down on the number of interrupts, leaving me one X100P for timming (so I don't need ztdummy.) MAYBE this would help 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 ___ 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] TDM-fxo card and zttest - logic probem?
Been playing around with zaptel/zttest utility and believe there is a logic problem with this 83 line app. (The objective is to better undertand missed frames, interrupts, etc, associated with the TDM card. Maybe we can get a handle on why things like spandsp failures, echo, etc, are occurring in some cases.) When the app is run as ./zttest -v, it repeatedly shows: 8192 samples in 8190 sample intervals 99.975586% 8192 samples in 8190 sample intervals 99.975586% 8192 samples in 8190 sample intervals 99.975586% 8192 samples in 8190 sample intervals 99.975586% implying the pci structure is running at about 99.975% accuarcy. Following the logic in the app, that really says the TDM card transfered 8,192 bytes of data in the equivalent timeframe as what 8,190 bytes would have been moved. In other words, we got the expected/wanted 8,192 bytes in 99.975% of the 1 second interval (indicating a better then expected response, not worse). Second, there is a rounding error in the calculation. In the statement: ms = (now.tv_sec - start.tv_sec) * 8000; the number of 'seconds' is calculated for the time necessary to receive something greater then 8,000 bytes from the TDM card. That statement always results in 1 second (times 8000 bytes per second to convert it into equivaltent byte counts). The statement that immediately follows it: ms += (now.tv_usec - start.tv_usec) / 125; calculates the number of 'microseconds' (in addition to the seconds from above), required to receive something greater then 8,000 bytes from the TDM card. On my system, that result is 23,863 microseconds. When converted to the equivalent number of bytes, it is 190.9 bytes. Adding the two values together results in 8,190.9 bytes, however the calculation drops everything to the right of the decimal (since the value is stuffed into an integer variable). Logic issues perceived include: 1. We received the expected 8,192 byes, period. There wasn't any missed frames that could be detected. The data in read in 1024 byte junks until something greater then 8,000 bytes (SIZE 8000) is received. One missed interrupt/frame is equivalent to 125,000 microseconds. So the measured 23,863 microseconds is far less then a single interrupt (125,000 microseconds), or about 19% of a single interrupt. (That would suggest a single missed interrupt (or frame) would yield a 91.02% result in the display. At what realistic percentage would problems arise? (It wouldn't appear that 99.975% is a serious problem, but what value is?) 2. On my system, the total accumlative time was 1.023863 seconds to receive the 8,192 bytes, when it was suppose to happen in 1.00 sec. If the app displayed those values, now we know what were looking for (23,863 usec of delay from something). 3. The entire zttest logic simply repeatedly reads data from the TDM buffer. There is no support in this app for interrupts, so if the interrupt service overhead (eg, scsi/ide/video delays) would impact how the interrupts were handled, it wouldn't be detected in the app logic. All we know is the time it took to get 8,192 bytes was something slightly greater then 1 second. Is the clock on the TDM card on frequency as an example? Who knows. So, that would imply the zttest app isn't just measuring bus/OS efficency, but includes all other imperfections including clock errors on the TDM board, and apparently excludes interrupt servicing. Another app is probably needed to narrow down the source of issues. 4. If we added logic to the app to simply read a TDM chip register as fast as it can, measure and report that, would that not provide some insight into how fast the pci bus and TDM card could respond (at max speed)? Can someone walk through the above and help me understand where my logic might be less then accurate/reasonable? Rich ___ 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 sound with voicemail and musiconhold?!?
Hi! I'am a new user and have problem with sound on a debian sarge. I can't play any sound with musiconhold or voicemail. Sounds on var/lib have good rights and mpg123 is installed. On console asterisk stops in the first playing. Someone have same problem or can help me? -- Antoine ___ 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] Demo phones with advertisement announcements
Craig wrote: There is a great missunderstanding between what you guys are talking now and what I want. I am not looking for advertiser to support free calls. However, I am asked now more often Can I test your service for a few days? These tests I have to pay from my own pocket. The customers should get the chance to test during different day / night time, but not a free service. Therefore they will get a limit of a certain amount what they can make phone calls for, the system should play some ads, but that ads would be more or less our own ads, hints, feature information, .. Another approach would be just let them make a phone call from 2 minutes, and not anymore the same number!!! bye Ronald A few years ago there was a company that set up in Australia offering free long distance calls, they played an add to you at the beginning and then every so often. Came out with a lot of fanfare and disappeared pretty quickly. Not sure if they went broke (most likely), couldn't find any users that wanted free phone calls (unlikely)or the concept was ahead of it's time (possibility). There was also a couple of people that pushed free dialup internet in return for users having to view a certain amount of their advertising. Unfortunately the business models didn't stack up and they went belly up. cr Message: 1 Date: Fri, 22 Apr 2005 08:06:57 +0800 From: Ronald Wiplinger [EMAIL PROTECTED] Subject: [Asterisk-Users] Demo phones with advertisement announcements To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I am asked very often to let the user try for a while. And of course they want to have it for free. However, there is nothing like free lunch out there. I got the idea to bother these people, by playing an advertisement before they actually make the call and even after a certain time. Has anybody done that before? Ideas I got for that is: 1. put the caller into a conference call with the advertisment channel 2. let the caller listen to the first advertisement before inviting the other party to the conference 3. keep playing ads, till the called party is in the conference too. 4. immediately silent the ads, when called party pickes up 5. wait the desired time and start to play the next advertisement block. Advanced feature: 1. give the caller the chance to pay for the call by key in a password(?), that means: a. it kicks out the advertisement or b. forward the call () so that the call is now directly connected. `?' means I am not sure if that is a good idea nor if that is possible bye Ronald ___ 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 -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ 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: Long incomingmsn line in capi.conf?
Hello, the incomingmsn line in chan_capi's capi.conf is limited to 80 characters (AST_MAX_EXTENSION default value). My problem: I have to include several MSNs but NOT all. The interface is a 30 channel PRI card with a number area of 600 numbers, splitted in different functions. Some numbers are used for fax, some for PPP, some for telephony. (Example: 1234567xx is used for fax, 1234568xx is used for ppp, 1234569xx is used for telephony) When I set incomingmsn to * it's fine for asterisk - it gets all calls - but PPP and fax are not working anymore because they don't get any calls. In Germany I have to take the whole number without the leading zero of the area prefix. So every MSN has a length of 10 characters. This limits the count of usable MSN to 7 (7*10 + 6 commas = 76 chars). I tried out to use a wildcard in the string (using the example above: 1234569*) but this doesn't work. Any idea (except modifying the source code)? Thank you! :-) Best regards Stefan ___ 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] Route SIP calls to provider
Your SIP provider doesn't need registration? Sounds good. Can you share the
IP address please?
Regards
Cameron
- Original Message -
From: iMRAN [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, April 21, 2005 4:35 AM
Subject: [Asterisk-Users] Route SIP calls to provider
Dear Pros,
Can anyone be kind enough to guide me to route calls to my SIP carrier.
I have configured * to as local PBX from Softphones to hardphones and
vice versa, the hardphone i have is AudioCodec MP108 8 FXS port
gateway.
SIP.conf
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=gsm
allow=g729
[1000]
type=friend
username=1000
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
[2000]
type=friend
username=2000
secret=password2
host=dynamic
context=internal
;canreinvite=yes
dtmfmode=rfc2833
extension.conf
[general]
static=yes
writeprotect=yes
[globals]
PHONE1=SIP/1000
PHONE2=SIP/2000
[international]
ignorepat = 88
exten= _1N1NXXNXX,1,Dial ???
[internal]
include = local-sip
[local-sip]
exten = 1000,1,Dial(${PHONE1},40,t)
exten = 1000,2,Hangup
exten = 2000,1,Dial(${PHONE2},40,t)
exten = 2000,2,Hangup
exten = 1001,1,Dial(${PHONE3},40,t)
exten = 1001,2,Hangup
i want user to dial 88 and they will get a tone and dial US or UK
number from local-sip context.
the provider only gave me a IP to route my SIP traffic and needs no
registration, can any please help me how to write the International
context in extension.conf also what i shld do in sip.conf..
my audiocodec supports g729 and g723 codec so do i need to aqquire
license for G729 from digium, if yes then why?
last if possible can you also please tell me wht i need to add on my
context so user can while in calling put the call on hold and transfer
to another sip phone.
I thankyou all for reading this mail and i hope someone will be kind
enough to help.
Best regards,
Imran
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RE: [Asterisk-Users] Mysql using Sip and voicemail
I can give you some help with the SIP stuff. Try it again with sip debug turned on and send the output back here. It would be good to see the SIP messages that are being transferred. The Asterisk SIP Stack is good, but not great. You might need to just add or delete an option in the sip.conf file. I did a very similar thing with SIP config and app_radius stuff. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Johnson Sent: Friday, April 22, 2005 8:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Mysql using Sip and voicemail I am currently running asterisk 1.0.7 and decided to try using MySQL to hold some of my voicemail and sip configuration. As a note - MySQL is already holding my CDR info. I followed the directions in on voip-info.org to copy files, modify Makefiles, recompile, and change the conf files accordingly. I have run into a few bumps that I need to ask about. With the voicemail database, if the voicemailbox is in the default context, all is well. If I attempt to place the mailbox in any other context, it will not work at all like the mailbox does not exist. The other issue is with the sip database, which does not to appear to work at all. When attempting to connect a SIP phone that setup in the database, I receive the error Registration from '999 sip:[EMAIL PROTECTED]' failed for '192.168.0.129' When starting asterisk, I can see where asterisk is logging into the MySQL database correctly. I have double checked the configuration files, database structure, and even tried setting the context as default (since that worked for voicemail) Any suggestions on what to check into Thanks Ben Johnson ___ 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] Grandstream : low bandwidth codec (ilbc doesn't work, any other ? )
Hi, I'm trying to setup one of free low bandwidth codecs for Grandstream (ilbc, g726, ...), but with ilbc I just hear engine running in handset. Is anyone using ilbc sucessfully with Grandstream? Quality ? Any other alternative ? I use Bristuffed Asterisk Thanks in advance, regards, Rob. ___ 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] SS7 for *
Hi!, Do you have a copy of the openss7 stack?? +*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*+ + Luciano Ramos+ + MCP - CCNA - CCNP (on the way :-)+ + Depto. de Internet, TelViso + + [EMAIL PROTECTED]+ + 02320-409125 + +*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*-*+ The box said 'Requires Windows 2000, NT, or better,' so I installed Linux. ___ 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] can't make my PRI dial out
On Fri, 22 Apr 2005 10:37:32 -0400 Mark Phillips [EMAIL PROTECTED] wrote: I have a full PRI installed on my * machine. I can get inbound calls just fine but can't make outbound ones. Zaptel.conf says; span=1,1,0,esf,b8zs bchan=1-23 dchan=24 zapata.conf says language=en context=default switchtype=4ess pridialplan=unknown signalling=pri_cpe channel=1-23 echocancel=yes group=1 Your zapata.conf should look like this: language=en context=default switchtype=4ess pridialplan=unknown signalling=pri_cpe echocancel=yes group=1 channel=1-23 You need to move the echocancel and the group above the channel line. The channel line definitions must be above and not below. 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] Recommendations for Spanish Voice Talent
We are putting together an IVR app that requires Spanish prompts. We're using Allison for the English prompts and are looking for recommendations for Spanish. Any thoughts? -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ 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] TE11OP - Mitel 200Sx??
I have done the same thing with an sx200 and a pri circuit
zaptel.conf
# t1 connected to the PRI circuit
span=1,1,0,exf,b8zs
# t1 connected to SX200
# the t1 card on my sx200 did d4 ami and I supplied ANI and DNIS through
the dial plan
span=2,0,0,d4,ami
bchan=1-23
dchan=24
em=25-47
-
zapata.conf
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=no
rxgain=0.0
txgain=0.0
useincomingcalleridonzaptransfer=yes
restrictcid=no
context=default
usecallingpres=yes
usercallerid=yes
hidecallerid=no
callerid=Company Name8005551212
signalling=pri_cpe
switchtype=dms100
group=1
channel = 1-23
group=2
signalling=em_w
emdigitwait=500
channel = 24-47
# I needed the emdigitwait=500 to wait long enough for the SX200 to dial
out it's digits
--
extensions.conf
# our PRI circiut gave us the last 4 digits of the dialed number and this
is how I passed
# *ANI*DNIS* to the SX200 for it to decode
# the first group were individual numbers that mapped to faxes and modems
exten = 8551,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = 8577,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = 8641,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = 8642,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = 1773,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = 1774,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
exten = 1775,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
# this set mapped our did 7000 - 7199 to the SX200
exten = _7[0-1]XX,1,Dial(Zap/g2/*${CALLERIDNUM}*${EXTEN}*,,r)
The reset of the dial plan took what ever I set up in the sx200 ARS to dial
out and
sent out put Zap/G1
Hope this helps
--
From: Henry Devito[SMTP:[EMAIL PROTECTED]
Reply To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, April 22, 2005 4:27 PM
To: Scott Wolfe; Asterisk Users Mailing List - Non-Commercial Discussion
Subject:Re: [Asterisk-Users] TE11OP - Mitel 200Sx??
File: ATT00231.htmlFile: ATT00232.txt
Don't you need one of these directives so the PRI knows which is master and
which is slave?
a.. pri_cpe: PRI signaling, CPE side
a.. pri_net: PRI signaling, Network side
Henry
- Original Message -
From: Scott Wolfe
To: Asterisk-Users@lists.digium.com
Sent: Friday, April 22, 2005 11:01 AM
Subject: [Asterisk-Users] TE11OP - Mitel 200Sx??
Hello all. I just received a TE110P and am trying to hook it to my Mitel
200SX has anyone successfully done this? My configuration is as follows.
Asterisk - TE110P -Kentrox (csu/dsu) - Mitel T1 Card.
All I get is a blinking yellow on my TE110P card and an alarm on my
Mitel. T1 card.
Any advice would be great.
Zaptel.conf
span=1,0,1,d4,ami
em=1-23
dchan=24
Zapata.conf
signalling=em_w
switchtype=dms100
echocancel=yes
echocancelwhenbridged=yes
echotraining=400
callerid=asreceived
group=1
context=default
channel = 1-23
--
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RE: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P?
Dell 1850 rack mount. We've been sourcing white box servers but can't beat Dell's price in the U.S. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Boehnlein Sent: Friday, April 22, 2005 2:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dell PowerEdge SC1425 w/ TE405P? On Fri, 22 Apr 2005, William Boehlke wrote: SC1425 is great value but note it does not have high availablility configurations. In our opinion, telephony requires dual NICs, dual power supplies and RAID 1 to have any hope of achieving five nines. William Boehlke What box would you reccomend for this? -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 4/21/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.10.2 - Release Date: 4/21/2005 ___ 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] Demo phones with advertisement announcements
Yep totally, same in Australia, pricing for long distance calls have crashed and changed the business case for this now and more so in the future. Voice advertising still does have it's place though - maybe in subsidising 'chat' rooms, with these you can definitely set up locally advertising access based on CID. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter http://www.0xdecafbad.com Sent: Friday, April 22, 2005 5:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Demo phones with advertisement announcements On Sat, 2005-04-23 at 07:56 +1130, Craig wrote: A few years ago there was a company that set up in Australia offering free long distance calls, they played an add to you at the beginning and then every so often. Came out with a lot of fanfare and disappeared pretty quickly. In america there was a company broadway that did the same thing. Aparently people didnt want to listen to a bunch of ads for 15 minutes (and it would interrorgate you with 'press number X' after each ad to make sure you werent just cueing minutes). When you place a call if the person didnt answer or it was busy you had to hang up and start all over, minutes were not saved. They did direct call the company doing the advertisement when listening to the ad if the person wanted to try buy the product. The idea seemed to be ok, although advertisers may not go for it since its hardly targeted. The implementation was horrible, if the call doesnt go through you should be able to try a different number. I knew a lot of people that would use that during idle time waiting for friends and what not to get home and would then call. In america now it cant sell well becuase mobile phones are incredibly cheap and most if not all offer unlimited long distance for $20 or less per month and unlimited nights and weekends. To compete with that would be difficult to say the least. There was also a couple of people that pushed free dialup internet in return for users having to view a certain amount of their advertising. Unfortunately the business models didn't stack up and they went belly up. In america free.org did dialup for free and got money based on access charges (the fees that phone companies pay each other any time calls go from one network to another). The advertisement based ISPs in america came later, and went away. No one wanted to see them to get what they could pay $10/mo to have without the ads. free.org went under once phone companies blacklisted their dialup numbers due to excessive fraud (I think figures put fraud at about 90% of all calls). Once no one could call, they had no revenue. The reasons that some of these companies went under were lack of planning on the user interface part, or lack of availability (free.org only had 1 city with numbers). I wouldnt imagine it would be hard to implement this in asterisk, however making it work so users are happy, lowering fraudulent callers (see recent litigation against google, yahoo, askjeeves, findwhat and others about fradulent clicks on ad banners), and other such things is the challenge. ATT also placed advertisements in their calling card plans that they gave soldiers in the middle east. They did this to avoid access charges (if its an 'informational service' there are no access charges) which the FCC didnt approve of. Basically a soldier would call into the system, hear a short ad and then their call would be placed. -- Trixter http://www.0xdecafbad.com UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 ___ 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] Recommendations for Spanish Voice Talent
We have recorded some prompts in Spanish by a male speaker, pls contact me offline for sending some of them if it suits your needs. Saludos / Regards Gustavo Russo - Original Message - From: George Pajari [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, April 22, 2005 6:25 PM Subject: [Asterisk-Users] Recommendations for Spanish Voice Talent We are putting together an IVR app that requires Spanish prompts. We're using Allison for the English prompts and are looking for recommendations for Spanish. Any thoughts? -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ 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: TE110p - universal voltage?
Thanks all. I too have found out that the card is both. Mike Tony Mountifield wrote: In article [EMAIL PROTECTED], Craig Guy [EMAIL PROTECTED] wrote: Can anyone with a TE110p confirm that it will fit and work in both a 3.3 and 5 volt pci slot? From photos it looks to be a universal card but the digium literature makes no mention of voltage requirements. I can cofirm that it has both the 5V and 3.3V cutouts in the edge connector. I can also confirm that I've used the card successfully in a 5V slot. I haven't tried it in the 3.3V slot. Cheers Tony ___ 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] can't make my PRI dial out
I haven't worked with PRI, but could it be related to an invalid callerid?
What about:
exten = _X., 1, SetCallerId(123123123)
exten = _X., 2, Dial(Zap/g1/${EXTEN})
Julian.
On 4/22/05, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On April 22, 2005 11:48 am, Mark Phillips wrote:
Nothing happens. I get the same (non)error.
I get plenty of output when receiving a call however.
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[Asterisk-Users] Re: can't make my PRI dial out
On Fri, Apr 22, 2005 at 11:10:43AM -0500, [EMAIL PROTECTED] wrote: I have a full PRI installed on my * machine. I can get inbound calls just fine but can't make outbound ones. If you run pri debug span x, you might see this behaviour: PRI debugging with the inbound numbers show that there is a minor difference in the SETUP frame: In the Channel ID, Telstra hasn't set the exclusive bit, AAPT has. Solution for me was in zapata.conf: - spanmap = 1,1,1 + spanmap = 1,1 Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] Demo phones with advertisement announcements
Yep, remember them well - I think the guy who was running it was called Paul Davies (could be wrong) - no idea if they are still running. I understand the big problem they had was securing advertising contracts, people under estimate the 'startup cost' in securing advertising (lol - something I'm currently trying to do for Australian IPTV) Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Sent: Friday, April 22, 2005 4:27 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Demo phones with advertisement announcements A few years ago there was a company that set up in Australia offering free long distance calls, they played an add to you at the beginning and then every so often. Came out with a lot of fanfare and disappeared pretty quickly. Not sure if they went broke (most likely), couldn't find any users that wanted free phone calls (unlikely)or the concept was ahead of it's time (possibility). There was also a couple of people that pushed free dialup internet in return for users having to view a certain amount of their advertising. Unfortunately the business models didn't stack up and they went belly up. cr Message: 1 Date: Fri, 22 Apr 2005 08:06:57 +0800 From: Ronald Wiplinger [EMAIL PROTECTED] Subject: [Asterisk-Users] Demo phones with advertisement announcements To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed I am asked very often to let the user try for a while. And of course they want to have it for free. However, there is nothing like free lunch out there. I got the idea to bother these people, by playing an advertisement before they actually make the call and even after a certain time. Has anybody done that before? Ideas I got for that is: 1. put the caller into a conference call with the advertisment channel 2. let the caller listen to the first advertisement before inviting the other party to the conference 3. keep playing ads, till the called party is in the conference too. 4. immediately silent the ads, when called party pickes up 5. wait the desired time and start to play the next advertisement block. Advanced feature: 1. give the caller the chance to pay for the call by key in a password(?), that means: a. it kicks out the advertisement or b. forward the call () so that the call is now directly connected. `?' means I am not sure if that is a good idea nor if that is possible bye Ronald ___ 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] can't make my PRI dial out
I made my zapata.conf look like the below (with relevant changes) and then programmed exten 3701 to dial my cell phone (I'm working remotely on this). I added the line exten = 3701,1,Dial(Zap/g1/19173657597) to extensions.conf and get this output from pri debug span 1 when I dial it -- Making new call for cr 32771 Protocol Discriminator: Q.931 (8) len=54 Call Ref: len= 2 (reference 3/0x3) (Originator) Message type: SETUP (5) [04 02 80 90] Bearer Capability (len= 4) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 0 User information layer 1: Unknown (24) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [28 0e b1 4d 61 72 6b 20 50 68 69 6c 6c 69 70 73] Display (len=14) Charset: 31 [ Mark Phillips ] [6c 08 21 83 32 32 32 32 30 38] Calling Number (len=10) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '17323571400' ] [70 0c 80 31 39 31 37 33 36 35 37 35 39 37] Called Number (len=14) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '19173657597' ] -- Called g1/19173657597 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 3/0x3) (Terminator) Message type: RELEASE COMPLETE (90) [08 02 83 e4] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Transit network (3) Ext: 1 Cause: Invalid information element contents (100), class = Protocol Error (6) ] -- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 1 got hangup NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null -- Hungup 'Zap/1-1' my call gets hung up immediatly. I know we are moving forward. I didn;t get this last time I tried to dial. Mark Andrew Kohlsmith wrote: On April 22, 2005 11:48 am, Mark Phillips wrote: Nothing happens. I get the same (non)error. I get plenty of output when receiving a call however. Odd... Here is my zapata.conf setup for my PRI: --- [channels] context=BellPRI switchtype=national pridialplan=unknown priindication=outofband overlapdial=no signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=no relaxdtmf=no group=1 callgroup=1 pickupgroup=1 immediate=no callerid=asreceived channel = 1-23 --- I simply Dial(Zap/g1/5551212). The fact that it doesn't think it can pick up a channel is interesting, I haven't run across that before. Combined with the fact that you can receive calls just fine, this is a very strange little problem. With my config (modified for your switchtype and context ONLY), what do you get when you try to dial a number? -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 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] Questions about a 7960 and images
Hello All, I was wondering how everyone got along with cisco 7960's. I just picked one up and I am having problems locating an image. I called cisco, but they will not sell to end users... Does anyone know a place where it can be purchased in the US? It has stock firmware, and the skinny seems to crash asterisk oddly... Also, does it require a download to run? For example, can I configure it, then bring it to another office and just plug in and go, or must it tftp from the head server? Regards, Greg ___ 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] voice pulse connect - no dtmf
I had the same problem with another provider whom I got no response from as usual.. We had 5 or 6 numbers that worked fine and one that just quit sending DTMF. - Original Message - From: Doug Harris To: [EMAIL PROTECTED] Digium. Com Sent: Friday, April 22, 2005 11:52 AM Subject: [Asterisk-Users] voice pulse connect - no dtmf Hi, I've got bunch of VP connect lines, and a day back two LA area numbers stop sending DTMF. They are IAX2. So, simply my customers can dial in, it hit my IVR but when they punch-in the number, my * running 1.0.7 cannot identify the dtmf. IAX debug does not show dtmf being sent to me. Just want to know whether any of you had this experience, and if so how that was fixed. Funny thing is this happened on two dids and others are OK. Cheers DH ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] can't make my PRI dial out
I know we are moving forward. I didn;t get this last time I tried to dial. Mark Why don't you try changing your switchtype to national from 4ess in your zapata.conf ___ 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] Line Noise UPDATE - If you've got line noise, read this
Since this thread has been going on for awhile, I've forgotten whether anyone mentioned that at least some Sipura products shipped with a 10 millisecond rtp time. My spa3k was this way. I changed it to 20 milliseconds and reboot. Might just try that if the setting is avail to you. We have basically the same setup. My cards are on 5 and 7 as well and I've disabled EVERYTHING is the bios that is not necessary; USB, serial, parallel, ect. I would think that if it was an IRQ issue, the call wouldn't tank when I connected it on the card with it's own IRQ. I just got in my new Cisco 7940 a few minutes ago and when I get a powersupply for it, I'm going to remove the Sipura-841 and try this one out and see if maybe that doesn't fix the problem. I'm doubtful, but it seems that this POS sipura is the only thing that is REALLY differing from out configurations. I didn't install X when I setup this box, so I know it's not running, but it's good to know that it can cause problems. Thanks for the info, every bit helps. Paul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walt Reed Sent: Friday, April 22, 2005 05:41 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Line Noise UPDATE - If you've got line noise,read this On Fri, Apr 22, 2005 at 02:40:10AM -0500, Paul said: Ok, well I managed to fix the IRQ conflicts...well, sorta. I have two X100P cards in the system. One now has it's own interrupt and the other is sharing one with the soundcard. I tested outbound calls on both cards, still have the damn static. I am so sick of this. Is anyone else using X100P cards and NOT having this problem?? Yes. I can make a call from a POTS phone hooked up to a Cisco ATA 186, out one X100P to the PSTN, back in a second X100P, to a phone hooked up to the second port on the ATA186 with no noise, and no echo, and a pretty small delay (which you can hear with one handset in each ear.) I have disabled most of the on-board I/O such as parallel, serial, and extra USB controllers, and the X100's are on int 5 and 7, not shared with anything. Interrupts 10 and 11 have a bunch of stuff shared and are used by USB controllers, ethernet ports (one on each IRQ) video card, SCSI controller, and one unknown device (some special nVidia device.) This machine is also used as a firewall / gateway / email server but does NOT run X (which I hear can cause problems on some machines.) I've been running this configuration for about 9 months with virtually no problems in a SOHO environment including weekly 3-hour long conference calls. I realize this doesn't help you much, but it IS possible for the configuration to work. I have been thinking about getting a Sipura 3000 to add another FXS port and remove one X100P which would also cut down on the number of interrupts, leaving me one X100P for timming (so I don't need ztdummy.) MAYBE this would help 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 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quadbri bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ?
Hi, I have problem with Quadbri and bristuffed Asterisk - I guess this is only configuration trick. I'd like Asterisk to respond only to 1 number on BRI interface and do nothing on other. Right now, even if I leave out that number in incoming context, Asterisk takes out and rejects call as number is non existant. I'd like that Asterisk wouldn't respond, so other ISDN phone I think a ugly trick is to do: exten = MSN_TO_BE_FREE,1,Wait(100) Matteo ___ 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] Upgrade Cisco 7940/7960 firmware
For the archive: http://www.cisco.com/univercd/cc/td/doc/product/voice/c_ipphon/english/ipp7960/addprot/mgcp/frmwrup.htm#wp1048832 ___ 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] QOS Routers
How about a linksys wrt54g with sveasoft firmware? Has some shaping and many other nice features... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Max Clark Sent: Friday, April 22, 2005 1:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] QOS Routers Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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 ___ 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: QOS Routers
Maybe this fits the bill. http://www.gigafast.com/products/product_detail/EE2400-SS.htm It retails for less than $100 Message: 9 Date: Fri, 22 Apr 2005 10:42:20 -0700 From: Max Clark [EMAIL PROTECTED] Subject: [Asterisk-Users] QOS Routers To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Hi all, I am looking for good (sub $200 dollars) routers to support VoIP installations. What is available at this point? I've used Netscreen and Checkpoint in the past, they are just too much overkill for this application. TIA, Max ___ 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
