[asterisk-users] Decrease group counter without hangup?
Hi, I have several parts of my dialplan implemented wirh the group function and now I need to decrease the group counter in some special case without hanging up the channel. Is this psossible ? So far I didn´t find something related in the documentation... Any hints will be appreciated... Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] A couple of questions for the Mitelgurus(phone-related - not systems)
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Barry Porch Sent: Friday, May 11, 2007 8:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] A couple of questions for the Mitelgurus(phone-related - not systems) Nigel, You cannot upgrade a non-dual mode 5220 to SIP. If you are referring to the cable that connects the 5310 to a 5235, that is a standard CAT5 straight-through cable. Barry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nigel Kendrick Sent: Friday, May 11, 2007 7:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] A couple of questions for the Mitel gurus(phone-related - not systems) Hi Folks, Just in case there are any Mitel gurus here: 1) Is it possible to convince a non-dual mode 5220 phone to 'upgrade' to the SIP firmware? I have inherited one that's Minet only. 2) I have a 5310 conference unit and 5235 phone in SIP mode, but someone's lost the connecting lead. Can anyone recommend anywhere in the UK for a replacement lead or confirm the pin-out so I can check whether a generic RJ-RJ lead will work without frying anything. Thanks Nigel Kendrick Thanks Barry, I managed to find some specs for the 5310 and when it mentioned that it was PoE powered I took the plunge and tried a cat 5 patch lead and everything's working fine. Shame about the 5220 - but no big loss. Nigel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 build system.
Hi, I have written an asterisk application app_custom.c and I want to link it with a third party library libthirdparty.so. Is there a way to do this with the 1.4 build system. Also does anyone have any documentation on customizing the 1.4 build system it's a lot different from the 1.2 build system and the make files seem cryptic. Regards, Rohan Hathiwala. DISCLAIMER == This e-mail may contain privileged and confidential information which is the property of Persistent Systems Pvt. Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Pvt. Ltd. does not accept any liability for virus infected mails. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quadbri and bristuff : no answer to isdn setup message
Hi, I'm trying to install a Junghanns quadbri for a few days but i stay with an asterisk error. (Everyone is busy/congested ) Asterisk is working with a Fritz PCbut from one year and now i want to add the quadbri. The quadbri card has been configured in NT mode and with no 100 ohms S/T termoination. (I'm not sure if the S/T parameter is correct) I have installed the bristuff package 0.4.0-test with : Zaptel-1.4.1 Libpri-1.4.0 Asterisk-1.4.2 The three package has been patched thank's bristuff package before compiling. When i try to use the channel ZAP/g1 my isdn equipment refuse to answer the SETUP message when i make : Exten = 1,Dial(ZAP/g1/069304993,20,rt) Thank's for your help. Here's my asterisk debug console log and my config : *CLI dial 1 == Console is full duplex -- Executing [EMAIL PROTECTED]:1] Set(OSS/dsp, CALLERID(all)=069670335) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(OSS/dsp, ZAP/g1/069671586|25|rt) in new stack 1 -- Making new call for cr 134 -- Requested transfer capability: 0x00 - SPEECH 1 nintranet*CLI [ 02 ff 03 08 01 06 05 04 03 80 90 a3 18 01 89 6c 0a 21 81 36 39 36 37 30 33 33 35 70 0a c1 30 36 39 36 37 31 35 38 36 a1 7d 02 91 81 ] 1 nintranet*CLI Unnumbered frame: 1 SAPI: 00 C/R: 1 EA: 0 TEI: 127EA: 1 1M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 41 bytes of data 1 Protocol Discriminator: Q.931 (8) len=41 1 Call Ref: len= 1 (reference 6/0x6) (Originator) 1 Message type: SETUP (5) 1 [04 03 80 90 a3] 1 Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) 1 Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) 1 Ext: 1 User information layer 1: A-Law (35) 1 [18 01 89]I 1 Channel ID (len= 3) [ Ext: 1 IntID: Implicit Other Spare: 0 Exclusive Dchan: 0 1 ChanSel: B1 channel 1 ] 1 [6c 0a 21 81 36 39 36 37 30 33 33 35] 1 Calling Number (len=12) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) 1Presentation: Presentation permitted, user number passed network screening (1) '69670335' ] 1 [70 0a c1 30 36 39 36 37 31 35 38 36] 1 Called Number (len=12) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '069671586' ] 1 [a1]net*CLI 1 Sending Complete (len= 1) 1 [7d 02 91 81] 1 High-layer compatibilty (len= 4) [ 1 0x91 1 0x81 1 ] 1 q931.c:3628 q931_setup: call 134 on channel 1 enters state 1 (Call Initiated) -- Called g1/069671586 1 No response to SETUP message 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending -- Channel 0/1, span 1 got hangup, cause 18 [May 14 11:10:49] DEBUG[25375]: chan_zap.c:3222 zt_setoption: Set option AUDIO MODE, value: ON(1) on Zap/1-1 [May 14 11:10:49] DEBUG[25375]: chan_zap.c:2674 zt_hangup: Hangup: channel: 1 index = 0, normal = 12, callwait = -1, thirdcall = -1 [May 14 11:10:49] DEBUG[25375]: chan_zap.c:2815 zt_hangup: Already hungup... Calling hangup once, and clearing call 1 NEW_HANGUP DEBUG: Destroying the call, ourstate Call Initiated, peerstate Overlap sending 1 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated, peerstate Overlap sending [May 14 11:10:49] DEBUG[25375]: chan_zap.c:1650 zt_disable_ec: disabled echo cancellation on channel 1 [May 14 11:10:49] DEBUG[25375]: chan_zap.c:3139 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/1-1 [May 14 11:10:49] DEBUG[25375]: chan_zap.c:1583 update_conf: Updated conferencing on 1, with 0 conference users [May 14 11:10:49] DEBUG[25375]: chan_zap.c:3218 zt_setoption: Set option AUDIO MODE, value: OFF(0) on Zap/1-1 [May 14 11:10:49] DEBUG[25375]: chan_zap.c:1650 zt_disable_ec: disabled echo cancellation on channel 1 -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'OSS/dsp' status is 'CHANUNAVAIL' cat /proc/zap/* Span 1: ztqoz/2/1 quadBRI PCI ISDN Card 1 Span 1 [NT] (cardID 7) Layer 1 DEACTIVATED (G2) AMI/CCS 1 ztqoz2/1/1 Clear (In use) 2 ztqoz2/1/2 Clear (In use) 3 ztqoz2/1/3 HDLCFCS (In use) Span 2: ztqoz/2/2 quadBRI PCI ISDN Card 1 Span 2 [NT] (cardID 7) Layer 1 DEACTIVATED (G2) AMI/CCS 4 ztqoz2/2/1 Clear (In use) 5 ztqoz2/2/2 Clear (In use) 6 ztqoz2/2/3 HDLCFCS (In use) Span 3: ztqoz/2/3 quadBRI PCI ISDN Card 1 Span 3 [NT] (cardID 7) Layer 1 DEACTIVATED (G2) AMI/CCS 7 ztqoz2/3/1 Clear (In use) 8 ztqoz2/3/2 Clear (In use) 9 ztqoz2/3/3 HDLCFCS (In use) Span 4: ztqoz/2/4 quadBRI PCI ISDN Card 1 Span 4 [NT] (cardID 7) Layer 1 DEACTIVATED (G2) AMI/CCS 10 ztqoz2/4/1 Clear (In use) 11 ztqoz2/4/2 Clear (In use) 12 ztqoz2/4/3 HDLCFCS (In use) Lspci 00:00.0 Host bridge: Intel Corporation 82865G/PE/P DRAM
Re: [asterisk-users] Dry Copper Pair
Am Freitag, den 11.05.2007, 18:44 -0400 schrieb Jon Pounder: just out of curiousity - anyone ever hijack pairs and get away with it ? (do your own cross connects on the street and utilize some crossconnect all within one branch of F1 cable out of the CO ?) I've been tempted in the past, and know that at least around here I would probably get away with it for quite some time before anyone actually cared enough to investigate. I know a setup where some kind of cable hijacking took place. A small office nearby had a regular ISDN/BRI line, with a small PBX and two or three analogue phones on it locally. One of the PBX extensions was connected to another telco copper pair (small houses usually have between 2 and about 16 pairs into the basement, depending on when the cable was laid), which led back to the Telco and came out at the office chief's home second copper pair, 30 meters away, other side of the street. I have been told that his predecessor had a son working for the then-German (in the 80s) Post office ;-) Pityfully this solution disappeared when a truck ran down the telco switchbox while reversing, must have been in 2002 or 2003. Just disconnected that 1 meter tall grey cupboard from the ground, leaving lots of cables dangling and while recabling, they (telco) obviously did not care to reconnect that special local exchange. The solution was to buy a DECT repeater and wireless handset, works like a charm in this situation. I have to admit I did not research this in full, but I assume that tampering with the Post property would have given you a night in jail - or left you without a job or pension plan, in this case, if you were caught. With deregulation, this probably softened a bit. Nowadays my impression is that the repair technicians are out-sourced service guys with a too-tight schedule - do not touch anything that is not necessary to be touched, because that might take time to be fixed or cost reimbursements or whatever. Documentation of cabling is existant (well, we are in Germany, after all): as it seems, it has been filed in a basement cupboard, locked, in an unused toilet room, guarded by wild dogs... at least service people tend to not have access to line whereabouts documentation, and no intention to ask too much questions. If the installation works, they just let it be. BTW this reminded me of these two BOFH episodes: http://www.theregister.co.uk/2005/02/25/bofh_2005_episode_7/ http://www.theregister.co.uk/2000/11/13/bofh_lights_out_for_contractors/ BR Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: C F wrote: Stephen i disagree. growing up in new work city i can say its quite easy to get away with it in the city. where i live now in new jersey (population of around 6) i wouldnt be able to pull that off. The world is a big place, and I suppose there's room for all kinds. In these parts, the vigilance is pretty high. The pillars are padlocked now; they didn't use to be, and the COs are locked down like Fort Knox. Anyway, I know enough more than one person who has landed in the clink for treating the telco like a personal lab. what exactly was the charge ? Perhaps something along the lines of unauthorised tampering with a telecomms installation? /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dialplan: execute on hangup
hi list,
I'm looking for a way to execute commands in my dialplan specifically when a
caller has hung up. my curretn dialplan looks like this:
exten = s,1,Answer
exten = s,n(restart),BackGround(intro)
exten = s,n,Read(Enter,4,4)
exten = s,n,Voicemail(${Enter},u)
exten = s,n,agi(process.php|${Enter})
exten = #,1,Playback(thanks)
exten = #,n,Hangup
it lets a user record a message to a chosen voicebox, when the user finishes
his message my pressing #, the script process.php is executed via AGI
everything fine. however, when the user finishes recording by simply hanging
up, asterisk isn't executing the AGI-command any more. what I'm looking for
is some kind of on hangup-hook to catch this use-case...
any help appreciated, thx!
michael
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Re: [asterisk-users] Dry Copper Pair
Per Jessen wrote: Perhaps something along the lines of unauthorised tampering with a telecomms installation? More likely conspiracy to aid terrorists by destroying the infrastructure. -- Chris Mason (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 UK 44.207.183.0271 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] function_db_read: DB requires an argument, DB(family/key)
from extensions.conf:
exten = _X.,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})})
I basically try to lookup the CLIP and attach a name for each inbound
call. This works fine, except when I have just restarted asterisk - at
which time I've more than once seen the message from the subject.
As far as I can tell, with my Set(CALLERID), I should always have an
argument in the DB function?
/Per Jessen, Zürich
--
ENIDAN Technologies GmbH - managed email security.
Starting at SFr1/month/user - http://www.spamchek.ch/
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[asterisk-users] Re: dialplan: execute on hangup
In article [EMAIL PROTECTED],
Michael Kamleitner [EMAIL PROTECTED] wrote:
I'm looking for a way to execute commands in my dialplan specifically when a
caller has hung up. my curretn dialplan looks like this:
exten = s,1,Answer
exten = s,n(restart),BackGround(intro)
exten = s,n,Read(Enter,4,4)
exten = s,n,Voicemail(${Enter},u)
exten = s,n,agi(process.php|${Enter})
exten = #,1,Playback(thanks)
exten = #,n,Hangup
it lets a user record a message to a chosen voicebox, when the user finishes
his message my pressing #, the script process.php is executed via AGI
everything fine. however, when the user finishes recording by simply hanging
up, asterisk isn't executing the AGI-command any more. what I'm looking for
is some kind of on hangup-hook to catch this use-case...
any help appreciated, thx!
Try something like:
exten = h,1,DeadAGI(process.php|${Enter})
Be prepared to experiment a bit!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] queue_exec: Unable to join queue
I have a queue defined which I use like this: exten = _X.(reception),n,Ringing exten = _X.,n,Queue(enidan,t,,,20) exten = _X.,n,Voicemail(443,u) exten = _X.,n,Hangup() When I start asterisk, this queue doesn't work - -- Executing [EMAIL PROTECTED]:3] Queue(mISDN/3-u0, enidan|t|||20) in new stack [May 14 13:53:59] WARNING[17860]: app_queue.c:3541 queue_exec: Unable to join queue 'enidan' -- Executing [EMAIL PROTECTED]:4] VoiceMail(mISDN/3-u0, 443|u) in new stack But all I need to do to fix it is reload app_queue. Does anyone know what's going on? /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call to Skype network
Open source...I wish...at least not to my knowledge yet. Likely something to do with the licensing for Skype...someone correct me here if appropriate. I'll drop a second email with details on the configuration unless someone else pipes up requesting it. D. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Verscheure Sent: Sunday, May 13, 2007 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call to Skype network yeah that would be great! Aren't there any open-source projects out there who handle this? greetz 2007/5/13, Dave Bour [EMAIL PROTECTED]: On x86 asterisk systems, there's 3 options out there, of which the Chanskype one I've found to be the best. It's $20 US for a single channel personal license or $99 / per channel on a business license. On the FreePBX systems/Trixbox, Tim Hunt wrote an excellent script to configure it. I've made a couple notes too if you want, I can send offlist (unless it's generally wanted here onlist as I don't like taking credit for others work). D -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Verscheure Sent: Saturday, May 12, 2007 9:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call to Skype network Hi everyone, Is it possible to call from your Asterisk server to the Skype network? i.e., let's say I would like to call from an extension from my Asterisk PBX machine to a Skype account, is this possible? I did a little bit of searching and they were talking about that's only possible with windows machines, is this true? greetz, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
Anybody I am still waiting. Kapil Dhawan wrote: Hi Can somebody brief me the working of RTP mixer from MeetMe perspective. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to bring MoH volume down
Hi, MoH volume is uncomfortably high and I want to bring it down. Its mpg123. How can I do it? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: dialplan: execute on hangup
thx Tony, but DeadAGI doesn't seem to fit my needs... the way I understand
its functioniality (
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI), DeadAGI
is ensureing that an executed AGI-script is finished, even if the caller
hung up _during_ execution.
in my case, I need to execute the AGI-script _after_ the user hung up the
voicemail is recorded.
another ideas: is there away to tell the Voicemail-command to execute an
AGI-script when recording is finsihed?
michael
On 5/14/07, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Michael Kamleitner [EMAIL PROTECTED] wrote:
I'm looking for a way to execute commands in my dialplan specifically
when a
caller has hung up. my curretn dialplan looks like this:
exten = s,1,Answer
exten = s,n(restart),BackGround(intro)
exten = s,n,Read(Enter,4,4)
exten = s,n,Voicemail(${Enter},u)
exten = s,n,agi(process.php|${Enter})
exten = #,1,Playback(thanks)
exten = #,n,Hangup
it lets a user record a message to a chosen voicebox, when the user
finishes
his message my pressing #, the script process.php is executed via AGI
everything fine. however, when the user finishes recording by simply
hanging
up, asterisk isn't executing the AGI-command any more. what I'm looking
for
is some kind of on hangup-hook to catch this use-case...
any help appreciated, thx!
Try something like:
exten = h,1,DeadAGI(process.php|${Enter})
Be prepared to experiment a bit!
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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--
Mag. Michael Kamleitner
-
[EMAIL PROTECTED]
https://www.xing.com/profile/Michael_Kamleitner
-
m-otion GmbH
Favoritenstr 4-6/III, 1040 Wien
+43 1 205705 / 21 (Fax 99)
-
www.m-otion.com
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RE: [asterisk-users] Dundi and unknown remote peers
Hmm, I tried this, but I get the following notice: NOTICE[27486]: pbx_dundi.c:4695 set_config: Ignoring invalid EID entry '*' Do you perhaps know for any other option? Thanks, Alex -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Bagnall Sent: Friday, May 11, 2007 7:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Dundi and unknown remote peers Is it possible to allow remote peers to connect to your local DUNDi Asterisk box, even if you don't have them listed in the dundi.conf? I seem to remember something in the sample config file about a [*] entry being possible... One would assume that would cover connections from undefined DUNDi clients. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited For full contact details visit http://www.minotaur.it/chris.html This email is made from 100% recycled electrons ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call to Skype network
Hi There, Good guide on setting up chanskype on trixbox http://www.geek-pages.com/articles/asterisk/setting_up_trixbox/asterisk_to_use_skype.html also: http://www.chanskype.com/ working on my trixbox 2.0 :) Best Regards, Com os melhores cumprimentos, Hugo Picão Link Consulting - RedesSegurança Tel: 213 100 182 Av. Duque de Ávila, 23 1000-138 Lisboa -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Bour Sent: segunda-feira, 14 de Maio de 2007 12:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Call to Skype network Open source...I wish...at least not to my knowledge yet. Likely something to do with the licensing for Skype...someone correct me here if appropriate. I'll drop a second email with details on the configuration unless someone else pipes up requesting it. D. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Verscheure Sent: Sunday, May 13, 2007 5:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Call to Skype network yeah that would be great! Aren't there any open-source projects out there who handle this? greetz 2007/5/13, Dave Bour [EMAIL PROTECTED]: On x86 asterisk systems, there's 3 options out there, of which the Chanskype one I've found to be the best. It's $20 US for a single channel personal license or $99 / per channel on a business license. On the FreePBX systems/Trixbox, Tim Hunt wrote an excellent script to configure it. I've made a couple notes too if you want, I can send offlist (unless it's generally wanted here onlist as I don't like taking credit for others work). D -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tim Verscheure Sent: Saturday, May 12, 2007 9:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Call to Skype network Hi everyone, Is it possible to call from your Asterisk server to the Skype network? i.e., let's say I would like to call from an extension from my Asterisk PBX machine to a Skype account, is this possible? I did a little bit of searching and they were talking about that's only possible with windows machines, is this true? greetz, Tim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil Dhawan wrote: Hi Can somebody brief me the working of RTP mixer from MeetMe perspective. (RTP? in MeetMe? I don't really know this well but AFAIK RTP doesn't get mixed directly in MeetMe. MeetMe is a generic Asterisk channels mixer) Aparantly people either don't know enough or don't have the time. Try rephrasing your question so it will be more specific and thus also hopefully take shorter time to answer. Do you have a working system? Do you need to set up one? What version of Asterisk? What types of channels do you try to mix? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Sudden appearance of SIP/2.0 401 Unauthorized
Did you have the IP specified in sip.conf? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Yaakov Menken Sent: May 13, 2007 10:43 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sudden appearance of SIP/2.0 401 Unauthorized Yesterday we moved one of our servers to a new IP. We updated DNS and various adapters configured to register to that server registered to the new IP correctly. All seemed to be well. This evening I discovered that with one exception, all of the adapters are getting a SIP/2.0 401 Unauthorized message back from asterisk. The exception is an Innomedia adapter -- Linksys PAP2's and (I believe) one Cisco ATA-188 are getting the Unauthorized. I have stopped, restarted, unloaded loaded sip, and erased astdb to start from scratch... no dice. None of the config files have changed, and, as I said, they all appeared to work last night. Can anyone give me a clue here? Yours, Yaakov Menken -- Yaakov Menken Capalon Communications, Inc. Ask us about Voice over IP for Business! http://www.capalon.com 888-CAPALON (227-2566) 410-358-9800 x120 410-510-1053 fax 443-413-1042 cell [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with queue
Asterisk 1.2.17 I am starting to have problem with one of my queue. Everytime when I try to login an agent with AgentCallBackLogin(), it will play periodic announcement for the queue during this function call. Also when this agent answer the call, during the conversation, the agent also hear the periodic announcement. I tried to delete the agent completely from the queue or recreate the queue, the problem still persist. I have not yet restart the asterisk because this is our production server. Gary ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bring MoH volume down
Remix your wav/mp3 files with a lower volume :) On 5/14/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Hi, MoH volume is uncomfortably high and I want to bring it down. Its mpg123. How can I do it? -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bring MoH volume down
Zeeshan, On Mon, 14 May 2007, Zeeshan Zakaria said something to this effect: MoH volume is uncomfortably high and I want to bring it down. Its mpg123. How can I do it? There are some settings in musiconhold.conf that may yield the desired effect: [default] mode=mp3 directory=/var/lib/asterisk/moh ; valid mode options: ; quietmp3 -- default ; mp3 -- loud ; mp3nb -- unbuffered ; quietmp3nb-- quiet unbuffered ; custom-- run a custom application ; files -- read files from a directory in any Asterisk supported format If not, it may come down to adjusting the base amplitude of the entire track down. I don't think there's a way to modify the gain specifically for MOH. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: RE: Digital Phones
We use the Handset Gateways from Citel. They convert SIP to Digital Handsets, so there is no hardware to add to the server and you can still use your 2-wire phone lines. -- -- Steven http://www.glimasoutheast.org bilal ghayyad [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi; Well, I understood now that Nortel has some digital phones that can be used with astrisk, but the question: what are the card models that should be installed on Asterisk server? Digium? What these models? Regards Bilal Ghayad Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Mobile Number to Mobile carrier mapping
Not now that they have intoduced number portability. The phone companies have to keep huge databases to keep track of which carrier to send the call to. -- -- Steven http://www.glimasoutheast.org Ritesh Agrawal [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and unicall + mfcr2 signalling
Hi, I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P card. I have one E1 with MFCR2 Signaling. I compiled asterisk + libunicall, and I can make calls over E1, but can't receive. Here the CLI when I make a call: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/23-081cbc40, Unicall/g1/91642208|50) in new stack -- Called g1/91642208 [May 14 10:34:59] NOTICE[4620]: chan_unicall.c:2599 handle_uc_event: Unicall/1 event Dialing [May 14 10:34:59] NOTICE[4620]: chan_unicall.c:1959 unicall_exception: Exception on 8, channel 1 [May 14 10:35:14] NOTICE[4620]: chan_unicall.c:2599 handle_uc_event: Unicall/1 event Alerting -- Hungup 'UniCall/1-1' == Spawn extension (ps5, 006191642208, 1) exited non-zero on 'SIP/23-081cbc40' Really destroying SIP dialog '[EMAIL PROTECTED]' Method: ACK [May 14 10:35:17] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event: Unicall/1 event Drop call [May 14 10:35:17] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event: Unicall/1 event Release call -- Unicall/1 released And here the CLI when I receive a call: [May 14 10:35:51] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event: Unicall/8 event Detected [May 14 10:35:52] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event: Unicall/8 event Protocol failure [May 14 10:35:52] ERROR[2914]: chan_unicall.c:2603 handle_uc_event: Unicall/8 protocol error. Cause 32772 Any idea why I can't receive calls, and fot Unicall protocol error Cause 32772? Thanks, Joca Loco. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Play a file on a channel from the Manager API
Is there any way to play a file on a channel from the Manager API (other than from Originate)? This question was asked by someone else on the ast-dev list and the only advice given was that Redirect was the solution. I find myself with the same problem now but I don't understand the response. The situation: I need to play a file from the Asterisk Manager on a channel that is currently in a call. I don't want to break them out of the call to play the message and I only want one specific channel to hear the message. In effect I want to ChanSpy the channel but to play a message instead of speak to the person on the channel. How does Redirect provide a solution? Thanks again, George ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: CITEL gateway does it work well?
The Citel Handset Gateways were the best option for our scenario. The cost per port for the number of buttons on our NEC DTerm/E phones was about half. Also, no network reengineering. We connected new 66 blocks to the Citel units. And just cutover from the old to the new. When you configure the extension on on the Citel, it does not register to asterisk until a phone is connected. Someday, I will upgrade our network, but using the old phones is much, much simpler at this point. Also, the way they did their BLFs is very asterisk freindly. We have been very happy with our units. -- -- Steven http://www.glimasoutheast.org Robert Augustyn [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi all, Is using a Citel gateway with Asterisk a good solution for reusing of the old Nortel digital phones? Would love to get some input from actual users. Any/all opinions welcome. robert -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT ? Number portability, land line to Cell
Having had various issues with local vendor (begins with V). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? From what I've read, the Fed's seem to say no, they cannot refuse, or impede this. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codename Pineapple - Chan_sip3 - what's the status?
Friends, I have gotten a few questions lately on the status on the Codename Pineapple project, the project that hopefully will produce a more stable and SIP compliant SIP stack for Asterisk. Due to lack of funding, it's postponed until further notice. I have a few sponsors, but not enough to be able to dedicate time to work on it. And since Digium hasn't made up their minds after thinking about it for more than a year, recent changes has not been updated on svn.digium.com The work that has been done so far, to mention some major issues - New configuration parser - New device type: phone (no more peers/users) - New way to handle messages (much less copying of in-memory data) - New transaction engine started - Adjustable SIP timers - Split into multiple source code files - Call pickup support - New registration handling This work has been sponsored by Edvina and Voop. Also, a lot of general cleanup and a new abstraction to prepare handling multiple sockets and domain-level configurations. I've gone through and changed quite a lot of the source. The Codename Pineapple SIP stack is already far away from chan_sip.c, but not anywhere close to something I would start testing. Work in progress or a sad ruin... You choose. I can't resist working on it now and then, but don't expect any major progress. If you have ideas on how to get the community to help fund a major overhaul like this, please send me e-mail off list. To find out more about Codename Pineapple, please visit http://www.codename-pineapple.org A big Thank You to Voop, Nuvio, TransNexus and Peter Gradwell for your support! Best regards, /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bring MoH volume down
Here the problem is that it is streaming audio from the Internet and I can't lower its volume. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT ? Number portability, land line to Cell
On Mon, 14 May 2007, Joe acquisto said something to this effect: Having had various issues with local vendor (begins with V). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? LNP does provide for this, at least in principle. -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT ? Number portability, land line to Cell
Quoting Joe acquisto [EMAIL PROTECTED]: Having had various issues with local vendor (begins with V). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? From what I've read, the Fed's seem to say no, they cannot refuse, or impede this. In Canada I've had limited success - personal lines seem easier to get done than business lines for some reason, and the whole issue of the overlapped area codes in the toronto area complicate things further (really or just portrayed that way as an excuse I am not really sure.) joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT ? Number portability, land line to Cell
I was going to port a number here in Ohio and Verizon said it would cost $90 to do so as they can charge what it cost them. Bob R Joe acquisto wrote: Having had various issues with local vendor (begins with V). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? From what I've read, the Fed's seem to say no, they cannot refuse, or impede this. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] zaptel huge irq problem
Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with BIOS PCI options, change priorities, PCI latencies, IRQ balance, smp_afinity, but impossible to come up with anything correcting that problem. Any idea about this? Is it possible to force the timer to ztdummy (RTC timer) when you have a zap card plugged in? It's the only thing i could try to make it work. Thanks, François. Just in case: - Linux 2.6.18 with debian patches and xen enabled, asterisk running on dom0. - Here is my zttest results under a bit of load: # ./zttest Opened pseudo zap interface, measuring accuracy... 99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062% 99.121094% 99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469% 99.414062% 99.902344% 99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406% 98.449707% 100.00% - The card DOES NOT seem to share interrupts (checked also with lspci): # cat /proc/interrupts CPU0 CPU1 1: 1626 0Phys-irq i8042 6: 3 0Phys-irq floppy 8: 0 0Phys-irq rtc 9: 0 0Phys-irq acpi 14: 63 0Phys-irq ide0 16: 1 0Phys-irq libata, eth3 17:6762583 0Phys-irq libata 18: 13789 0Phys-irq libata 19: 33459690 0Phys-irq eth1 20: 19864325 0Phys-irq sky2, eth0 21: 269250881 0Phys-irq wctdm 256: 77735119 0 Dynamic-irq timer0 257:3986325 0 Dynamic-irq resched0 258: 37 0 Dynamic-irq callfunc0 259: 04652748 Dynamic-irq resched1 260: 0139 Dynamic-irq callfunc1 261: 0 28924306 Dynamic-irq timer1 262: 1021 0 Dynamic-irq xenbus 263: 0 0 Dynamic-irq console NMI: 0 0 LOC: 0 0 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Remote extensions not working on provider's wireless Internet connection
Hi Zeeshan, On 5/13/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I've solved this problem. It was very easy (only if I knew how to do it before). I changed the UDP ports, i.e. 1. In sip.conf, bindport=5070 2. In my IP Phone server settings, www.myserver.com:5070 Now it seems to be working good and I hope there'll be no more problem with it. Sorry for not replying earlier; I got your note late, and then when I woke up had no Internet. Ah, the joys of Rogers. I'm glad to hear you solved it -- my only concern would be if you now want to connect ordinary 5060 looking phones. I will do a bit of research, I'm sure Asterisk can bind to more then one port. Thanks, Gerald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT ? Number portability, land line to Cell
Joe acquisto wrote: Having had various issues with local vendor (begins with V). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? From what I've read, the Fed's seem to say no, they cannot refuse, or impede this. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users It depends on whether or not your local vendor actually owns the number itself or acquires it from another vendor. If they own the number, I don't think they're legally allowed to refuse to port it. If they lease the number from another provider, they're not actually obligated to assist you in porting, although they might, if you ask nicely, tell you with whom you need to deal to port the number. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk High-Capacity Stability
On May 12, 2007, at 4:11 PM, Atlanticnynex wrote: Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on dimensioning and it seems theres some mixed feelings about Asterisk in high-capacity environments. I guess I'm looking for input as to whether Asterisk could handle roughly one DS3's worth of calls (672 calls) just doing the LCR (I've seen some pre-built LCR apps, looks like they all do on-the-fly MySQL queries- I think I'd write my own AGI that would use a cache). With my hardware, could Asterisk run stable for this amount of traffic? What stability issues does Asterisk have at this scale? Simply put, NO. I am on a project now where a client had an OpenSER box acting as an SBC and registrar passing traffic to several asterisk boxes which are doing LCR lookups on the fly as well as writing custom CDRs all through PHP AGI scripts to a Postgres DB. The Asterisk boxes do not scale, and randomly start swallowing calls or, more often, restart the process (safe_asterisk is handling this). There is some light IVR type usage for reporting account balances and the like. With anything more than 80 or 90 calls on the box, the IVR prompts start to break up. Ben through replacing hardware, more memory, different Asterisk builds, etc. I've had an open issue with Digium support on this for at least a couple of weeks, and the best advice so far was try using the SVN build. That makes things better, but it's still not anywhere close to fixed.. It's absolutely incredible that Asterisk works at all for some of the situations its been put in - major kudos to the developers. But I don't think using it for what you're talking about is a long-term business strategy. When the highlight of the 1.6 release is bridging channels, you know high volume sip to sip usage in a carrier class call routing environment is NOT what development is focused on. And that's fine. If you use a wrench to do the job of a screwdriver, you shouldn't complain when you bust your knuckles That being said, I don't meant to trash Asterisk at all. It's a fantastic feature server, and a great PBX, both of which things I use it for very successfully. I just don't think it's ready to handle 50k plus minutes a day SIP to SIP with LCR and billing data, no matter what you do with it. I'm 100% positive there are people out there doing it successfully, but those are the exception, not the rule. And I doubt they are running unmodified code. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How obtain the slot position when a call is parked?
Hi, I want to ask you if asterisk, when I use the command park(), gives me for example a variable that contains the slot position where it parks the call or if it only tells me (audio) in the channel this position number? In other words, is there a way to obtain and use the value of the slot position when the call is parked? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bring MoH volume down
Please provide us with your config in musiconhold.conf so I/we can see how you are streaming. There may be a way to lower the volume, but it depends on how you are performing the streaming. On 5/14/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: Here the problem is that it is streaming audio from the Internet and I can't lower its volume. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT ? Number portability, land line to Cell
Having had various issues with local vendor (begins with V). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? From what I've read, the Fed's seem to say no, they cannot refuse, or impede this. Your local Vendor can certainly refuse to port the number, regardless of whether or not they're actually supposed to allow portability. They're the phone company, they don't have to care. Excuses can range fom we don't support that to the equipment's too old to my dog ate my homework. They know that 99.9% of all consumers are stupid and/or will not argue the point. Most people do not choose to engage big businesses over things like this. That's unfortunate, of course, because it enables companies to get away with blowoffs like this successfully and makes it harder for the rest of us to fight. You might find it interesting and/or useful to see if you can get them to port it to their own wireless division, assuming that they have one. If you decide to press the point, which you're encouraged to do, then the following resource ought to be helpful. http://www.fcc.gov/cgb/consumerfacts/numbport.html ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanSpy
Hi Guys, Does anyone know if is it possible to put one channel in two different spygroups? Thanks! Alex ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk High-Capacity Stability
On Mon, 14 May 2007, Daryl Jurbala said something to this effect: That being said, I don't meant to trash Asterisk at all. It's a fantastic feature server, and a great PBX, both of which things I use it for very successfully. Agreed. And, it's worth pointing out, that's what Asterisk is intended to be at this point; it's an *endpoint*, a UA. Excellent as a feature server, voicemail depository, PBX, IVR, what have you, *not* as a router or a PC-host based softswitch. About the only possible use I could imagine for such a thing in a routing scenario is as a broker that commands superior intelligence and is able to use extensive logic in call decisions (like LCR) and then releases itself from the media and signaling path entirely, but if you want that, you can't really use a B2BUA, and a SIP proxy like OpenSER can do that much better. Fast. Because that's what it's designed to do. I am not sure why so many people want to use it in call routing scenarios, because it's not a transit system. That's what optimised network elements are for; media gateways, proxies, etc. -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How is Context Determined when Transferring a Call?
When trasferring a call, how is the context determined? When using a zap device, and the DTMF code for blind or attended transfer is entered, does the tranfer originate at the context the zap device is set to be in, or does it originate from where the outside call being transferred originated in, or the context the current call is in? I ask because I am seeing strange behavior when trying to transfer some calls placed on zap devices. Dial plan logic does goto's to get the call to a context, i.e., internal or PSTN contexts. Some transfers fail though, because, say the call was originally to the PSTN context, and a transfer fails if it is to an internal context. Instead of goto's, is it best to use macros? Or a bunch of include statements? Sincerely, Brent A. Torrenga Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 tel:+1 219 836 8918 x325 fax:+1 219 836 1138 email:[EMAIL PROTECTED] web:www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] OT ? Number portability, land line to Cell
I think Joe's analysis is unreasonably negative regarding the landline companies' willingness to port. The link he provides, http://www.fcc.gov/cgb/consumerfacts/numbport.html, reflects my experience. A couple cautions, however: Landline companies may take two to three weeks to actually complete the port (as the FCC says, DO NOT cancel your current service until the new service is actually working). Your new carrier will request an LOA (Letter of Authorization) to complete the port. Make sure that the LOA is limited to making changes only to the service that you want them to be changing and the account title (for your existing service), service address, account number, etc., are exactly correct on the LOA. Otherwise you'll hear from your new carrier in a couple weeks that the old carrier refuses to complete the port because the existing customer is ABC Enterprises and the new customer is A. B. Cooper Enterprises. (This is why they may request a copy of your existing phone bill--to make sure everything is letter-perfect.) --Don Don Kelly PCF Corp Real Support for your Virtual Office 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Greco Sent: Monday, May 14, 2007 9:59 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] OT ? Number portability, land line to Cell Having had various issues with local vendor (begins with V). am looking to move to all wireless. Anyone know if current vendor can refuse to port the current land line numbers to a wireless provider? From what I've read, the Fed's seem to say no, they cannot refuse, or impede this. Your local Vendor can certainly refuse to port the number, regardless of whether or not they're actually supposed to allow portability. They're the phone company, they don't have to care. Excuses can range fom we don't support that to the equipment's too old to my dog ate my homework. They know that 99.9% of all consumers are stupid and/or will not argue the point. Most people do not choose to engage big businesses over things like this. That's unfortunate, of course, because it enables companies to get away with blowoffs like this successfully and makes it harder for the rest of us to fight. You might find it interesting and/or useful to see if you can get them to port it to their own wireless division, assuming that they have one. If you decide to press the point, which you're encouraged to do, then the following resource ought to be helpful. http://www.fcc.gov/cgb/consumerfacts/numbport.html ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] zaptel huge irq problem
François, I too had a similar problem and found the information on this page helpful: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting What ended up working for me was changing the UDMA to mode 2 for the hard drive. Once I did that, this card has worked perfectly for me. Michael L. Young -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Monday, May 14, 2007 10:24 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] zaptel huge irq problem Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with BIOS PCI options, change priorities, PCI latencies, IRQ balance, smp_afinity, but impossible to come up with anything correcting that problem. Any idea about this? Is it possible to force the timer to ztdummy (RTC timer) when you have a zap card plugged in? It's the only thing i could try to make it work. Thanks, François. Just in case: - Linux 2.6.18 with debian patches and xen enabled, asterisk running on dom0. - Here is my zttest results under a bit of load: # ./zttest Opened pseudo zap interface, measuring accuracy... 99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062% 99.121094% 99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469% 99.414062% 99.902344% 99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406% 98.449707% 100.00% - The card DOES NOT seem to share interrupts (checked also with lspci): # cat /proc/interrupts CPU0 CPU1 1: 1626 0Phys-irq i8042 6: 3 0Phys-irq floppy 8: 0 0Phys-irq rtc 9: 0 0Phys-irq acpi 14: 63 0Phys-irq ide0 16: 1 0Phys-irq libata, eth3 17:6762583 0Phys-irq libata 18: 13789 0Phys-irq libata 19: 33459690 0Phys-irq eth1 20: 19864325 0Phys-irq sky2, eth0 21: 269250881 0Phys-irq wctdm 256: 77735119 0 Dynamic-irq timer0 257:3986325 0 Dynamic-irq resched0 258: 37 0 Dynamic-irq callfunc0 259: 04652748 Dynamic-irq resched1 260: 0139 Dynamic-irq callfunc1 261: 0 28924306 Dynamic-irq timer1 262: 1021 0 Dynamic-irq xenbus 263: 0 0 Dynamic-irq console NMI: 0 0 LOC: 0 0 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
I was reading an article on RTP Mixer so started studying about the mixing done by Asterisk in MeetMe. Read that CC should contain the no of participants ifupto 15 and CSRC should come, but not getting any by asterisk. Tzafrir Cohen wrote: On Sat, May 12, 2007 at 10:44:19AM +0530, Kapil Dhawan wrote: Hi Can somebody brief me the working of RTP mixer from MeetMe perspective. (RTP? in MeetMe? I don't really know this well but AFAIK RTP doesn't get mixed directly in MeetMe. MeetMe is a generic Asterisk channels mixer) Aparantly people either don't know enough or don't have the time. Try rephrasing your question so it will be more specific and thus also hopefully take shorter time to answer. Do you have a working system? Do you need to set up one? What version of Asterisk? What types of channels do you try to mix? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simultaneous Capacity
Hi List I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is it feasible. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How obtain the slot position when a call is parked?
On Monday 14 May 2007 10:41 am, [EMAIL PROTECTED] wrote:
I want to ask you if asterisk, when I use the command park(), gives me for
example a variable that contains the slot position where it parks the call
or if it only tells me (audio) in the channel this position number? In
other words, is there a way to obtain and use the value of the slot
position when the call is parked? Thanks.
No. You need to use ParkAndAnnounce and a feature I'd managed to get added
which lets you get the parking slot number in the dialplan variable
${PARKEDAT}.
I use it like this:
exten = _X.,1,...
exten = _X.,n,ParkAndAnnounce(PARKED,,Local/[EMAIL PROTECTED])
...
[parkinginfo]
exten = s,1,NoOp(PARKEDAT=${PARKEDAT})
exten = s,n,...
So basically when the call gets parked, it announces the parking slot to a
Local channel which executes in [parkinginfo]. Parkinginfo can write it to a
db, SMS it to a skywriter, whatever you want.
It'd be nice to get this sent to a SIP phone and make parking just that much
more useful, but as the guys say... patches welcome. :-)
-A.
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[asterisk-users] Difference between making a call and Originate
When I make a regular call from my SIP phone connected to my Asterisk server I have no issues, however when I make a call using Originate : 'Channel'=SIP/[EMAIL PROTECTED], 'Context'='mycontext', 'Exten'='899', 'Priority'=1, 'Callerid'='whatever')); It creates a screech sound when the first audio file is played. Doesn't seem to happen with another VSP I tried, but still, why would a regular outbound call work just fine and Originate create this strange sound. I know for sure that it isn't the audio file that I'm playing by the way. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Capacity
On Mon, 14 May 2007, Kapil Dhawan said something to this effect: I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is it feasible. In truth, it is very unlikely. How are you planning to pick up the PRIs, anyway? 3 quad-span T1 cards? -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Call to Skype network
Here's my instructions...based off Tim Hunt's great script...needs
cleanup but the gist is hear to get someone going...you may think I'm
reboot happy as there's more than a couple here but past experience
found that reloads didn't do it...reboot seem to get things
going...probably something simple...haven't taken time to resolve.
#get gcc, qt-devel, kernel-devel, asterisk-devel if not installed
#check if security patches applied, if kernel-devel and asterisk-devel
done via source...don't yum install those.
yum install gcc qt-devel kernel-devel.i686 asterisk-devel
# download and install the timhunt chanskype installation...run the
installer file
wget http://www.timhunt.net/stuff/chanskype/chanskype-trixbox.tgz
# run makeaccount.sh 1 (or however many accounts you want to make)
./makeaccount.sh 1
#remove twm and install ratpoison per notes from ChanSkype
yum remove twm
wget
http://savannah.nongnu.org/download/ratpoison/ratpoison-1.4.1.tar.gz
tar -zxf rattab
cd ratptab
./configure
make
make install
# create autostart for vncserver using ntsysv
ntsysv
#find vncserver and mark it for autostart.
#reboot
reboot
# download latest release of chanskype
# located at ftp://ftp.chanskype.com/download/packages/
# find the redhat enterprise version (usually FC3), ie ends with
FC3-RHEL4.bin
# ie... chanskype-1.2.9-FC3-RHEL4.bin
wget ftp://ftp.chanskype.com/download/packages/chanskype*FC3-RHEL4.bin
chmod +x chanskype*
./chanskype-x.x.x.-FC3-RHEL4.bin
# where x.x.x is the release you downloaded.
#if it fails and running trixbox repeat with 1DOT2 after the command
./chanskype-x.x.x.-FC3-RHEL4.bin 1DOT2
# access vnc for your server for each port and log into skype with your
account info.
# ie my server is 192.168.101.150:1 (first port assigned for skype1
account)
# move license if exists to proper location
mkdir /var/lib/instant
mkdir /var/lib/instant/licenses
cp ~/CS* /var/lib/instant/licenses
reboot
edit /etc/asterisk/skype.conf for number of channels in use (same as the
number from makeaccount).
nano /etc/asterisk/skype.conf
open freepbx...create custom trunk...add dial plan as desired...preface
any dial with 00+ ie..
001+nxxnxx
00+1nxxnxx
then under outgoing settings, custom dial string... add:
Local/[EMAIL PROTECTED]
using config editor...add following 3 lines to extensions_custom
[skype]
exten = _X.,1,Dial(Skype/any/${EXTEN})
exten = _X.,2,Hangup
#save settings.
#reboot
reboot
#using vnc ...for each account created, assign user account in skype.
Save settings...
# now the catch that makes Chanskype annoying... Resolve the prompt that
makes skype prompt you to allow chanskype access otherwise you fail to
get a channel assigned.
#cd to your account created for skype...ie skype1...
cd .skype
cd to your login account on skype
edit config.xml and find line Authorizations/Authorizations..change
to Authorizationschan_skype/Authorizations
#reboot for last time...
reboot
#you are ready to go. Log into your box, and connect to the asterisk
session and check your channels...should show as 1 (or however many you
purchased) licensed channels.
/usr/sbin/asterisk -rvx skype status
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Re: [asterisk-users] Dry Copper Pair
Per Jessen wrote: Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: C F wrote: Stephen i disagree. growing up in new work city i can say its quite easy to get away with it in the city. where i live now in new jersey (population of around 6) i wouldnt be able to pull that off. The world is a big place, and I suppose there's room for all kinds. In these parts, the vigilance is pretty high. The pillars are padlocked now; they didn't use to be, and the COs are locked down like Fort Knox. Anyway, I know enough more than one person who has landed in the clink for treating the telco like a personal lab. what exactly was the charge ? Perhaps something along the lines of unauthorised tampering with a telecomms installation? I wasn't going to bother replying to Jon's post, because, well, some things aren't worth the bother. But here it is, for the public good. First, there's section 326 of the Criminal Code of Canada: Theft of telecommunication service 326. (1) Every one commits theft who fraudulently, maliciously, or without colour of right, (a) abstracts, consumes or uses electricity or gas or causes it to be wasted or diverted; or (b) uses any telecommunication facility or obtains any telecommunication service. Then, there's section 334: Punishment for theft 334. Except where otherwise provided by law, every one who commits theft (a) is guilty of an indictable offence and liable to imprisonment for a term not exceeding ten years, where the property stolen is a testamentary instrument or the value of what is stolen exceeds five thousand dollars; or (b) is guilty (i) of an indictable offence and is liable to imprisonment for a term not exceeding two years, or (ii) of an offence punishable on summary conviction, where the value of what is stolen does not exceed five thousand dollars. The person in question was slapped with a $10,000 fine. Look, these guys take tampering with wire infrastructure seriously. There's a reason the addresses aren't published, the buildings non-descript, and the doors locked nine ways to Sunday. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
Kapil Dhawan wrote: I was reading an article on RTP Mixer so started studying about the mixing done by Asterisk in MeetMe. Read that CC should contain the no of participants ifupto 15 and CSRC should come, but not getting any by asterisk. I'll just leave it at this so we can all move on with our lives: Asterisk isn't totally an RTP Mixer in the sense you are reading about. It is an audio mixer. Frame of audio comes in over RTP, gets sent in (only the audio portion) to be mixed, frame comes out, gets turned into RTP again. The RTP part has no idea that multiple sources were mixed together, 'nor should it care. The sources could have been Zaptel channels for example in which case they couldn't be added to the list. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bring MoH volume down
Alex Balashov wrote: Zeeshan, On Mon, 14 May 2007, Zeeshan Zakaria said something to this effect: MoH volume is uncomfortably high and I want to bring it down. Its mpg123. How can I do it? There are some settings in musiconhold.conf that may yield the desired effect: [default] mode=mp3 directory=/var/lib/asterisk/moh ; valid mode options: ; quietmp3 -- default ; mp3 -- loud ; mp3nb -- unbuffered ; quietmp3nb-- quiet unbuffered ; custom-- run a custom application ; files -- read files from a directory in any Asterisk supported format If not, it may come down to adjusting the base amplitude of the entire track down. I don't think there's a way to modify the gain specifically for MOH. Is there a way to do it for voice mail messages? I have a user who has trouble hearing the voice messages, saying they are too quiet. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
Thanks Michael, I've already been through all that unfortunately, and I have a SATA drive, so no UDMA mode 2 as far as I know. I'm currently trying everything again anyway, but i doubt it will work if nothing worked the first time. Anyone would know of issues with XEN or SMP (or both) kernel? Do dual core AMD64 processors have issues? François. Michael L. Young wrote: François, I too had a similar problem and found the information on this page helpful: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting What ended up working for me was changing the UDMA to mode 2 for the hard drive. Once I did that, this card has worked perfectly for me. Michael L. Young -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Monday, May 14, 2007 10:24 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] zaptel huge irq problem Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with BIOS PCI options, change priorities, PCI latencies, IRQ balance, smp_afinity, but impossible to come up with anything correcting that problem. Any idea about this? Is it possible to force the timer to ztdummy (RTC timer) when you have a zap card plugged in? It's the only thing i could try to make it work. Thanks, François. Just in case: - Linux 2.6.18 with debian patches and xen enabled, asterisk running on dom0. - Here is my zttest results under a bit of load: # ./zttest Opened pseudo zap interface, measuring accuracy... 99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062% 99.121094% 99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469% 99.414062% 99.902344% 99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406% 98.449707% 100.00% - The card DOES NOT seem to share interrupts (checked also with lspci): # cat /proc/interrupts CPU0 CPU1 1: 1626 0Phys-irq i8042 6: 3 0Phys-irq floppy 8: 0 0Phys-irq rtc 9: 0 0Phys-irq acpi 14: 63 0Phys-irq ide0 16: 1 0Phys-irq libata, eth3 17:6762583 0Phys-irq libata 18: 13789 0Phys-irq libata 19: 33459690 0Phys-irq eth1 20: 19864325 0Phys-irq sky2, eth0 21: 269250881 0Phys-irq wctdm 256: 77735119 0 Dynamic-irq timer0 257:3986325 0 Dynamic-irq resched0 258: 37 0 Dynamic-irq callfunc0 259: 04652748 Dynamic-irq resched1 260: 0139 Dynamic-irq callfunc1 261: 0 28924306 Dynamic-irq timer1 262: 1021 0 Dynamic-irq xenbus 263: 0 0 Dynamic-irq console NMI: 0 0 LOC: 0 0 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ François Delawarde Ingeniero de red Tel: 918.03.92.51 E-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] _ WIRELESS MUNDI http://www.wirelessmundi.com/ C/Isaac Newton, 1 - Oficina 26 · Parque Tecnológico de Madrid 28760 TRES CANTOS (Madrid) Tlf./Fax: (+34) 918 03 92 51 La información contenida en este mensaje y en sus archivos adjuntos es CONFIDENCIAL y se dirige exclusivamente a sus destinatarios. Queda expresamente prohibida la utilización de la misma por cualquier persona distinta de los destinatarios de esta comunicación. Si usted ha recibido este mensaje por error le rogamos que lo comunique inmediatamente a WIRELESS MUNDI y lo borre al igual que todos sus documentos adjuntos. El correo electrónico no puede asegurar la confidencialidad ni la integridad de sus mensajes por lo que WIRELESS MUNDI no se hace responsable de tales errores u omisiones. --0-- All information in this message and its attachments is confidential and may be legally privileged. Only intended recipients are authorized to
Re: [asterisk-users] Re: CITEL gateway does it work well?
Steven wrote: The Citel Handset Gateways were the best option for our scenario. The cost per port for the number of buttons on our NEC DTerm/E phones was about half. Also, no network reengineering. I've noticed that all the people who have good things to say about them are using East Asian digital phones (e.g. NEC, Toshiba, Samsung, etc); the NEC phones I have worked with don't give DTMF feedback in a native install, so you wouldn't notice much difference. But the Nortel phones do (or are able to, depending on how the switch is configured); losing that ability would bother me. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OT (semi) E60 problem
Hello again gurus. I have been using Asterisk with great results going on a couple of years now. My primary box is running asterisk 1.42 built from a tar ball on Mac OSX 10.4.9. I have a very odd issue that I cannot seem to nail down, which is related to my Nokia E60 SIP phone. I use the E60 with very good results (latest firmware) from several locations. Basically it works fine from everywhere EXCEPT when it's on the same LAN as my asterisk box. The SIP config. that I have setup for the LAN connection refers to my asterisk box by it's local IP (ie 192.168.1.101). The external configs refer to the asterisk box by it's name (ie sip.domain.com). It seems like this has something to do with the authentication realm? If I create a new config on the phone using the LAN address, it works, but then when I leave the LAN, it appears to register, but issues a connection error when I try to place calls. To get it working again from outside the LAN, I can change the realm parameter in SIP.conf, and then reload and then change it AGAIN back to it's original value and reload, the phone then works fine. At that point, the LAN based config won't work anymore and will give me a connection error. Any thoughts on this? Ideas on how to troubleshoot further or work around it would be great. Thanks, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: TC400B load problem
On May 14, 2007, at 4:53 AM, Arun Kumar wrote: Im trying to install my TC400B trans coder card when I do: modprobe wctc4xxp tail -f /var/log/messages says: May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with 92 transcoders (srcs=000c, dsts=0101) May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with 92 transcoders (srcs=0101, dsts=000c) May 13 14:56:36 pbx2 kernel: Zaptel DTE (g.729a / g.723.1 5.3kbps) Transcoder support LOADED (firm ver = 56) May 13 14:56:49 pbx2 kernel: wctc4xxp: probe of :03:01.0 failed with error -5 That looks like a problem that you should talk with Digium Support about. Matthew Fredrickson ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
Hi, Francois: François Delawarde wrote: Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with BIOS PCI options, change priorities, PCI latencies, IRQ balance, smp_afinity, but impossible to come up with anything correcting that problem. What kind of motherboard do you have? - The card DOES NOT seem to share interrupts (checked also with lspci): # cat /proc/interrupts CPU0 CPU1 1: 1626 0Phys-irq i8042 6: 3 0Phys-irq floppy 8: 0 0Phys-irq rtc 9: 0 0Phys-irq acpi 14: 63 0Phys-irq ide0 16: 1 0Phys-irq libata, eth3 17:6762583 0Phys-irq libata 18: 13789 0Phys-irq libata 19: 33459690 0Phys-irq eth1 20: 19864325 0Phys-irq sky2, eth0 21: 269250881 0Phys-irq wctdm 256: 77735119 0 Dynamic-irq timer0 257:3986325 0 Dynamic-irq resched0 258: 37 0 Dynamic-irq callfunc0 259: 04652748 Dynamic-irq resched1 260: 0139 Dynamic-irq callfunc1 261: 0 28924306 Dynamic-irq timer1 262: 1021 0 Dynamic-irq xenbus I've never seen cat /proc/interrupts output that looks like that... waaaitaminute... are you running this in a virtual machine? Or on a machine running virtual machines? -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF not recognizing *
With our current setup, we have an older avaya system which is linked with our asterisk system via a em wink connection. When you press 2 on the avaya network, it will jump to our asterisk box and then sends DTMF digits. Asterisk listens for those numbers and then responses as soon as it has a match. The problem is with having a send to voicemail option. Right now, a user can press *5053 and they will be sent directly to that user's voicemail box, rather than their phone. But when you press 2*5053, it appears the * is ignored or not sent. I need to find a way to make this option work. Any thoughts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bring MoH volume down
On Mon, 14 May 2007, Stephen Bosch said something to this effect: Is there a way to do it for voice mail messages? I have a user who has trouble hearing the voice messages, saying they are too quiet. From a cursory glance at the voicemail settings, I can't see a way. The voicemail messages are stored in a fairly coherent directory structure and it may be possible to develop some sort of process that funnels the voicemail recordings through a volume-boosting 'sox' transformation or similar, specifically for that user, but other than that I can't think of anything off the top of my head. Volume issues tend to have to do with the phone the user is using as often as not. The user might benefit from a phone with decent volume controls/output gain boosting. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Simultaneous Capacity
Just a quick brief I have a requirement of running 10 PRI's (300 Channels). I still have to decide on hardware and cards. Can you suggest some. As per my understanding it will be tough to go beyond 150. Alex Balashov wrote: On Mon, 14 May 2007, Kapil Dhawan said something to this effect: I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is it feasible. In truth, it is very unlikely. How are you planning to pick up the PRIs, anyway? 3 quad-span T1 cards? -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk High-Capacity Stability
Several people do use it for handling 50k minutes a day. (I'm one of them). Yes, you need to know what you are doing, and have a nice design, but it is possible.Our code is only slightly altered. (mainly for billing purposes). Zoa Daryl Jurbala wrote: On May 12, 2007, at 4:11 PM, Atlanticnynex wrote: Thanks Alex, some great ideas. I think, however, I'm leaning towards Asterisk at this point- since I have quite a bit of experience there, and very little with SER. At this point, I'm wondering from a dimensioning standpoint, what kind of capacity my machine will have (Dual Core Xeon 2.4GHz 4GB RAM). As I said, I don't plan to do any transcoding. I read the voip-info page on dimensioning and it seems theres some mixed feelings about Asterisk in high-capacity environments. I guess I'm looking for input as to whether Asterisk could handle roughly one DS3's worth of calls (672 calls) just doing the LCR (I've seen some pre-built LCR apps, looks like they all do on-the-fly MySQL queries- I think I'd write my own AGI that would use a cache). With my hardware, could Asterisk run stable for this amount of traffic? What stability issues does Asterisk have at this scale? Simply put, NO. I am on a project now where a client had an OpenSER box acting as an SBC and registrar passing traffic to several asterisk boxes which are doing LCR lookups on the fly as well as writing custom CDRs all through PHP AGI scripts to a Postgres DB. The Asterisk boxes do not scale, and randomly start swallowing calls or, more often, restart the process (safe_asterisk is handling this). There is some light IVR type usage for reporting account balances and the like. With anything more than 80 or 90 calls on the box, the IVR prompts start to break up. Ben through replacing hardware, more memory, different Asterisk builds, etc. I've had an open issue with Digium support on this for at least a couple of weeks, and the best advice so far was try using the SVN build. That makes things better, but it's still not anywhere close to fixed.. It's absolutely incredible that Asterisk works at all for some of the situations its been put in - major kudos to the developers. But I don't think using it for what you're talking about is a long-term business strategy. When the highlight of the 1.6 release is bridging channels, you know high volume sip to sip usage in a carrier class call routing environment is NOT what development is focused on. And that's fine. If you use a wrench to do the job of a screwdriver, you shouldn't complain when you bust your knuckles That being said, I don't meant to trash Asterisk at all. It's a fantastic feature server, and a great PBX, both of which things I use it for very successfully. I just don't think it's ready to handle 50k plus minutes a day SIP to SIP with LCR and billing data, no matter what you do with it. I'm 100% positive there are people out there doing it successfully, but those are the exception, not the rule. And I doubt they are running unmodified code. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bring MoH volume down
Is there a way to do it for voice mail messages? I have a user who has trouble hearing the voice messages, saying they are too quiet. Just one user? Sounds like a user problem... however, with that said, you can try increasing your zaptel volumes. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP Mixer
Perfect Josh...but if i am running an application which has a capability of showing number or participants depending upon CC value, that doesn't work. Secondly, Asterisk can show on CLI about current talking users where it is maintaining talking status but not sending it down the line to be used by other apps. Anyways, i will go with your statement and leave it on core developers to comment. Joshua Colp wrote: Kapil Dhawan wrote: I was reading an article on RTP Mixer so started studying about the mixing done by Asterisk in MeetMe. Read that CC should contain the no of participants ifupto 15 and CSRC should come, but not getting any by asterisk. I'll just leave it at this so we can all move on with our lives: Asterisk isn't totally an RTP Mixer in the sense you are reading about. It is an audio mixer. Frame of audio comes in over RTP, gets sent in (only the audio portion) to be mixed, frame comes out, gets turned into RTP again. The RTP part has no idea that multiple sources were mixed together, 'nor should it care. The sources could have been Zaptel channels for example in which case they couldn't be added to the list. Joshua Colp Software Developer Digium, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 'Invalid characters in name' with asterisk-gui
This belongs in the asterisk-gui mailing list. However, I will see what I can do. -bkruse FYI. It is just a javascript pattern matching function, its super easy to change. Tom Lobato wrote: Hi all! Is there a way to asterisk-gui to allow underline (as such cpd_tom) in Names? It allows to [di]enable alphanumeric, but not underline noway. Why such restriction in asterisk-gui if even asterisk users.conf allows (and works fine) it? Thank you, Tom Lobato ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
Stephen Bosch wrote: Per Jessen wrote: Jon Pounder wrote: Quoting Stephen Bosch [EMAIL PROTECTED]: C F wrote: Stephen i disagree. growing up in new work city i can say its quite easy to get away with it in the city. where i live now in new jersey (population of around 6) i wouldnt be able to pull that off. The world is a big place, and I suppose there's room for all kinds. In these parts, the vigilance is pretty high. The pillars are padlocked now; they didn't use to be, and the COs are locked down like Fort Knox. Anyway, I know enough more than one person who has landed in the clink for treating the telco like a personal lab. what exactly was the charge ? Perhaps something along the lines of unauthorised tampering with a telecomms installation? I wasn't going to bother replying to Jon's post, because, well, some things aren't worth the bother. But here it is, for the public good. First, there's section 326 of the Criminal Code of Canada: Theft of telecommunication service 326. (1) Every one commits theft who fraudulently, maliciously, or without colour of right, (a) abstracts, consumes or uses electricity or gas or causes it to be wasted or diverted; or (b) uses any telecommunication facility or obtains any telecommunication service. Then, there's section 334: Punishment for theft 334. Except where otherwise provided by law, every one who commits theft (a) is guilty of an indictable offence and liable to imprisonment for a term not exceeding ten years, where the property stolen is a testamentary instrument or the value of what is stolen exceeds five thousand dollars; or (b) is guilty (i) of an indictable offence and is liable to imprisonment for a term not exceeding two years, or (ii) of an offence punishable on summary conviction, where the value of what is stolen does not exceed five thousand dollars. The person in question was slapped with a $10,000 fine. Look, these guys take tampering with wire infrastructure seriously. There's a reason the addresses aren't published, the buildings non-descript, and the doors locked nine ways to Sunday. I will add that utility lines usually have easements for the public and private land they run across. I signed easements for the overhead power line, the buried telco cable and the wiring pedestal. They run about 650 feet on my private driveway. They are on my property but climbing the poles, excavating near the cable or opening the pedestal are forms of trespass. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Double DTMF digits
On Sun, 2007-05-13 at 20:54 +0300, Dovid B wrote: I am actually getting DTMF over SIP when people call in to a clients system that is running a2billing. They are using RFC2833. If you are using a Cisco router anywhere in the loop, there is a known bug that causes rfc2833 and inband signalling to cause double DTMF. It is fixed in IOS 12.4.11T ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF not recognizing *
On Mon, 14 May 2007, Rob Schall said something to this effect: The problem is with having a send to voicemail option. Right now, a user can press *5053 and they will be sent directly to that user's voicemail box, rather than their phone. But when you press 2*5053, it appears the * is ignored or not sent. I need to find a way to make this option work. You would need some way of debugging what DTMF signals actually appear in the context of the bearing channels on the EM trunk. Is there any apparent difference with the duration of the tones? That is to say, how long one holds down the 2 or the *? Also, does it vary depending on the delay? Is it possible to press 2, wait 3-4 seconds, and then press * and see it work? -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
François Delawarde wrote: Thanks Michael, I've already been through all that unfortunately, and I have a SATA drive, so no UDMA mode 2 as far as I know. I'm currently trying everything again anyway, but i doubt it will work if nothing worked the first time. Anyone would know of issues with XEN or SMP (or both) kernel? Do dual core AMD64 processors have issues? Uh, yeah... Xen has many, many problems with interrupt handling and is utterly unsuitable for running anything that depends on hardware peripherals. I speak from very painful experience. There is no way, under any circumstance, that I would try to run Asterisk with interface cards in a Xen environment. It's too bad you wasted so much time trying to fix it -- it's never going to work. Try ripping Xen out and doing it directly on the physical server. I think you'll find your problems will go away. -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to bring MoH volume down
Alex Balashov wrote: On Mon, 14 May 2007, Stephen Bosch said something to this effect: Is there a way to do it for voice mail messages? I have a user who has trouble hearing the voice messages, saying they are too quiet. From a cursory glance at the voicemail settings, I can't see a way. The voicemail messages are stored in a fairly coherent directory structure and it may be possible to develop some sort of process that funnels the voicemail recordings through a volume-boosting 'sox' transformation or similar, specifically for that user, but other than that I can't think of anything off the top of my head. Volume issues tend to have to do with the phone the user is using as often as not. The user might benefit from a phone with decent volume controls/output gain boosting. Well, we've already cranked the hell out of his phone. He says that phone conversations are at a reasonable volume now, but it's the voicemail that's a problem. I guess we'd have to tweak the gain on the TDM card. (The user is in a rock band. :) ) -Stephen- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Areski CDR
Hi folks, I was wondering what happened to Areski CDR viewer that came before with Freepbx. I've noticed that the live-CD contains Areski but the repositories don't have it. Will you fix that? or shall I install Areski from sources? Regards, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User #382615 - http://counter.li.org/ SIP # 1-747-633-6676 Ext. 1011 FWD # 764839 Ext. 1011 http://routerman.blogsome.com http://gst.telecom.pucp.edu.pe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
Stephen Bosch wrote: # cat /proc/interrupts CPU0 CPU1 1: 1626 0Phys-irq i8042 6: 3 0Phys-irq floppy 8: 0 0Phys-irq rtc 9: 0 0Phys-irq acpi 14: 63 0Phys-irq ide0 16: 1 0Phys-irq libata, eth3 17:6762583 0Phys-irq libata 18: 13789 0Phys-irq libata 19: 33459690 0Phys-irq eth1 20: 19864325 0Phys-irq sky2, eth0 21: 269250881 0Phys-irq wctdm 256: 77735119 0 Dynamic-irq timer0 257:3986325 0 Dynamic-irq resched0 258: 37 0 Dynamic-irq callfunc0 259: 04652748 Dynamic-irq resched1 260: 0139 Dynamic-irq callfunc1 261: 0 28924306 Dynamic-irq timer1 262: 1021 0 Dynamic-irq xenbus I've never seen cat /proc/interrupts output that looks like that... waaaitaminute... are you running this in a virtual machine? Or on a machine running virtual machines? It looks like a XEN machine. Well spotted, Stephen. /Per Jessen, Zürich -- ENIDAN Technologies GmbH - managed email security. Starting at SFr1/month/user - http://www.spamchek.ch/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and unicall + mfcr2 signalling
try using testcall with 255 as debug level and report back results in order to be able to help you. http://www.moythreads.com/unicall/mfcr2-asterisk-unicall-0.2-english.pdf On 5/14/07, Joca Loco [EMAIL PROTECTED] wrote: Hi, I'm running Asterisk 1.4.2 on a Debian GNU/Linux and a Digium TE210P card. I have one E1 with MFCR2 Signaling. I compiled asterisk + libunicall, and I can make calls over E1, but can't receive. Here the CLI when I make a call: -- Executing [EMAIL PROTECTED]:1] Dial(SIP/23-081cbc40, Unicall/g1/91642208|50) in new stack -- Called g1/91642208 [May 14 10:34:59] NOTICE[4620]: chan_unicall.c:2599 handle_uc_event: Unicall/1 event Dialing [May 14 10:34:59] NOTICE[4620]: chan_unicall.c:1959 unicall_exception: Exception on 8, channel 1 [May 14 10:35:14] NOTICE[4620]: chan_unicall.c:2599 handle_uc_event: Unicall/1 event Alerting -- Hungup 'UniCall/1-1' == Spawn extension (ps5, 006191642208, 1) exited non-zero on 'SIP/23-081cbc40' Really destroying SIP dialog '[EMAIL PROTECTED]' Method: ACK [May 14 10:35:17] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event: Unicall/1 event Drop call [May 14 10:35:17] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event: Unicall/1 event Release call -- Unicall/1 released And here the CLI when I receive a call: [May 14 10:35:51] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event: Unicall/8 event Detected [May 14 10:35:52] NOTICE[2914]: chan_unicall.c:2599 handle_uc_event: Unicall/8 event Protocol failure [May 14 10:35:52] ERROR[2914]: chan_unicall.c:2603 handle_uc_event: Unicall/8 protocol error. Cause 32772 Any idea why I can't receive calls, and fot Unicall protocol error Cause 32772? Thanks, Joca Loco. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk High-Capacity Stability
Daryl Jurbala wrote: There is some light IVR type usage for reporting account balances and the like. With anything more than 80 or 90 calls on the box, the IVR prompts start to break up. Ben through replacing hardware, more memory, different Asterisk builds, etc. Zoa wrote: Several people do use it for handling 50k minutes a day. (I'm one of them). Yes, you need to know what you are doing, and have a nice design, but it is possible.Our code is only slightly altered. (mainly for billing purposes). Zoa, I've been experiencing the IVR prompts breaking up, as Daryl mentioned. The problem also affects queue announcements, but native music-on-hold sounds good. Have you experienced this problem, and if so what steps did you take to correct it? Thank you, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 peer unreachable in one direction - NAT problem?
The situation is one of my asterisk servers is behind a NAT firewall and one is not. Both servers have multiple IAX peers. The NAT firewall has port 4569 mapped through to the asterisk server behind. But, the natted server is almost permanently unreachable from this non-natted server, even though, the non-natted server is almost permanently _reachable_ from the natted server. Details are below with iax2 debug and core debug 3. I actually have an Asterisk 1.2 and an Asterisk 1.4 server in the non-natted role, and both have the same issue. However, I have another non-natted server (on a different ISP) that can talk fine to the natted server. (IP addresses replaced with names.) myNonNattedServer*CLI iax2 show peers Name/UsernameHost Mask Port Status myNattedServUN myNattedServer (S) 255.255.255.255 4569 (T) UNREACHABLE [May 14 19:06:05] DEBUG[5549]: chan_iax2.c:1154 update_max_nontrunk: New max nontrunk callno is 7 [May 14 19:06:05] DEBUG[5549]: chan_iax2.c:1252 find_callno: Creating new call structure 6 [May 14 19:06:05] DEBUG[5551]: chan_iax2.c:1644 send_packet: Sending 12 on 6/0 to myNattedServer:4569 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00012ms SCall: 6 DCall: 0 [myNattedServer:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00012ms SCall: 5 DCall: 6 [myNattedServer:37657] Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 5 [myNattedServer:37657] [May 14 19:06:05] DEBUG[5546]: chan_iax2.c:4788 raw_hangup: Raw Hangup myNattedServer:37657, src=6, dst=5 [May 14 19:06:06] DEBUG[5540]: chan_iax2.c:1644 send_packet: Sending 12 on 6/0 to myNattedServer:4569 Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00012ms SCall: 6 DCall: 0 [myNattedServer:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00012ms SCall: 6 DCall: 6 [myNattedServer:37657] Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 6 [myNattedServer:37657] [May 14 19:06:06] DEBUG[5542]: chan_iax2.c:4788 raw_hangup: Raw Hangup myNattedServer:37657, src=6, dst=6 myNattedServer*CLI iax2 show peers Name/UsernameHost Mask Port Status myNonNattedSeUN myNonNattedServ (S) 255.255.255.255 4569 (T) OK (14 ms) Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00016ms SCall: 00010 DCall: 0 [myNonNattedServ:4569] May 14 18:08:45 DEBUG[1196]: chan_iax2.c:1007 update_max_nontrunk: New max nontrunk callno is 12 May 14 18:08:45 DEBUG[1196]: chan_iax2.c:1112 find_callno: Creating new call structure 11 May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6654 socket_read: Received packet 0, (6, 30) May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6848 socket_read: IAX subclass 30 received May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6857 socket_read: For call=11, set last=16 May 14 18:08:45 DEBUG[1196]: chan_iax2.c:1515 send_packet: Sending 16 on 11/10 to myNonNattedServ:4569 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00016ms SCall: 00011 DCall: 00010 [myNonNattedServ:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 00010 DCall: 00011 [myNonNattedServ:4569] May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6654 socket_read: Received packet 0, (6, 10) May 14 18:08:45 DEBUG[1196]: chan_iax2.c:6848 socket_read: IAX subclass 10 received May 14 18:08:45 DEBUG[1196]: chan_iax2.c:7510 socket_read: Immediately destroying 11, having received INVAL May 14 18:08:45 DEBUG[1196]: chan_iax2.c:7513 socket_read: Destroying call 11 Also when calls are placed to myNonNattedServer from myNattedServer (which does work), the channel name is IAX2/myNattedServer:37657-callno, as opposed to IAX2/myNattedServUserName-53. (BTW, if I turn off qualify on myNonNattedServer, I can still not make calls from myNonNattedServer to myNattedServer.) Any idea what is wrong? This used to work fine (possibly when myNattedServer was only trying to talk to one asterisk server through the NAT - now it has 3, only one of which is working properly). Many thanks, Sebastian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dry Copper Pair
Stephen i disagree. growing up in new work city i can say its quite easy to get away with it in the city. where i live now in new jersey (population of around 6) i wouldnt be able to pull that off. The world is a big place, and I suppose there's room for all kinds. In these parts, the vigilance is pretty high. The pillars are padlocked now; they didn't use to be, and the COs are locked down like Fort Knox. Anyway, I know enough more than one person who has landed in the clink for treating the telco like a personal lab. what exactly was the charge ? Perhaps something along the lines of unauthorised tampering with a telecomms installation? I wasn't going to bother replying to Jon's post, because, well, some things aren't worth the bother. But here it is, for the public good. First, there's section 326 of the Criminal Code of Canada: Theft of telecommunication service 326. (1) Every one commits theft who fraudulently, maliciously, or without colour of right, (a) abstracts, consumes or uses electricity or gas or causes it to be wasted or diverted; or (b) uses any telecommunication facility or obtains any telecommunication service. Then, there's section 334: Punishment for theft 334. Except where otherwise provided by law, every one who commits theft (a) is guilty of an indictable offence and liable to imprisonment for a term not exceeding ten years, where the property stolen is a testamentary instrument or the value of what is stolen exceeds five thousand dollars; or (b) is guilty (i) of an indictable offence and is liable to imprisonment for a term not exceeding two years, or (ii) of an offence punishable on summary conviction, where the value of what is stolen does not exceed five thousand dollars. The person in question was slapped with a $10,000 fine. Look, these guys take tampering with wire infrastructure seriously. There's a reason the addresses aren't published, the buildings non-descript, and the doors locked nine ways to Sunday. In the neighborhood where I live in Putnam County NY, Verizon recently posted a sign for a $50,000 reward for information leading to the arrest of individuals responsible for tampering with their infrastructure. Apparently, someone had repeatedly hacked the same piece of equipment (don't know what it was - they wouldn't say). I don't know what the criminal codes say, but it is obviously an offense for which you can be arrested, and Verizon felt it was important enough to give away a sizeable sum to defend their equipment and access to their network. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] zaptel huge irq problem
Try switching to a Sangoma card. You wont have anymore IRQ issues once you abandon Digium hardware. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Monday, May 14, 2007 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] zaptel huge irq problem Thanks Michael, I've already been through all that unfortunately, and I have a SATA drive, so no UDMA mode 2 as far as I know. I'm currently trying everything again anyway, but i doubt it will work if nothing worked the first time. Anyone would know of issues with XEN or SMP (or both) kernel? Do dual core AMD64 processors have issues? François. Michael L. Young wrote: François, I too had a similar problem and found the information on this page helpful: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting What ended up working for me was changing the UDMA to mode 2 for the hard drive. Once I did that, this card has worked perfectly for me. Michael L. Young -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Monday, May 14, 2007 10:24 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] zaptel huge irq problem Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with BIOS PCI options, change priorities, PCI latencies, IRQ balance, smp_afinity, but impossible to come up with anything correcting that problem. Any idea about this? Is it possible to force the timer to ztdummy (RTC timer) when you have a zap card plugged in? It's the only thing i could try to make it work. Thanks, François. Just in case: - Linux 2.6.18 with debian patches and xen enabled, asterisk running on dom0. - Here is my zttest results under a bit of load: # ./zttest Opened pseudo zap interface, measuring accuracy... 99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062% 99.121094% 99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469% 99.414062% 99.902344% 99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406% 98.449707% 100.00% - The card DOES NOT seem to share interrupts (checked also with lspci): # cat /proc/interrupts CPU0 CPU1 1: 1626 0Phys-irq i8042 6: 3 0Phys-irq floppy 8: 0 0Phys-irq rtc 9: 0 0Phys-irq acpi 14: 63 0Phys-irq ide0 16: 1 0Phys-irq libata, eth3 17:6762583 0Phys-irq libata 18: 13789 0Phys-irq libata 19: 33459690 0Phys-irq eth1 20: 19864325 0Phys-irq sky2, eth0 21: 269250881 0Phys-irq wctdm 256: 77735119 0 Dynamic-irq timer0 257:3986325 0 Dynamic-irq resched0 258: 37 0 Dynamic-irq callfunc0 259: 04652748 Dynamic-irq resched1 260: 0139 Dynamic-irq callfunc1 261: 0 28924306 Dynamic-irq timer1 262: 1021 0 Dynamic-irq xenbus 263: 0 0 Dynamic-irq console NMI: 0 0 LOC: 0 0 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ François Delawarde Ingeniero de red Tel: 918.03.92.51 E-mail: [EMAIL PROTECTED] _ WIRELESS MUNDI http://www.wirelessmundi.com/ C/Isaac Newton, 1 - Oficina 26 · Parque Tecnológico de Madrid 28760 TRES CANTOS (Madrid) Tlf./Fax: (+34) 918 03 92 51 _ La información contenida en este mensaje y en sus archivos adjuntos es CONFIDENCIAL y se dirige exclusivamente a sus destinatarios. Queda expresamente prohibida la utilización de la misma por cualquier persona distinta de los destinatarios de esta
Re: [asterisk-users] Simultaneous Capacity
I have a requirement of running 10 PRI's (300 Channels). I still have to decide on hardware and cards. Can you suggest some. As per my understanding it will be tough to go beyond 150. Alex Balashov wrote: On Mon, 14 May 2007, Kapil Dhawan said something to this effect: I want to try Asterisk with 10 PRI on a single Xeon machine with g711. Is it feasible. In truth, it is very unlikely. How are you planning to pick up the PRIs, anyway? 3 quad-span T1 cards? You'll probably want to look into creating a Dundi cluster. I would not recommend putting more than two Quad span T1 cards in a single machine. Actually, I probably wouldn't put more than one in a single machine. If you use 3 Quad PRI cards each one in a different machine and configure those 3 machines as a Dundi cluster, you should be able to fulfill your 10PRI / 300 channel requirement. You might want to consider even five, six, or more machines. It would certainly reduce the impact of a hardware failure. - Noah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk High-Capacity Stability
On May 14, 2007, at 1:29 PM, Zoa wrote: Several people do use it for handling 50k minutes a day. (I'm one of them). Yes, you need to know what you are doing, and have a nice design, but it is possible.Our code is only slightly altered. (mainly for billing purposes). That's great if you're good enough/have the time to make that happen. But when I have issues and call/pay Digium and don't get timely or meaningful answers, it's doesn't make for a good business decision to continue using it for that purpose when I can toss in a Nextone or Sansay and have it just work. All the time. No babysitting. Full professional and timely problem resolution from the vendor, etc, etc, etc. Don't even get me started on Digium not being able to get TC400Bs to properly negotiate g.723.1 5.3k when a client requests 6.3k first (thank god for Cantata). I guess it all comes down to whether you want things to just work and be able to have tier 1/2 support capable of actually doing anything meaningful, or if you want to have the engineering level people forced to do all the work. From my standpoint, the smart business decision is quite clear. But, as I said, Asterisk is still driving the feature servers, and works well for it. As mentioned by someone else previously in the thread, it makes a great endpoint. If you're having good success with it, that's fantastic. I would hope that you contribute back to the list how you set things up to make this a possibility. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Remote extensions not working on provider's wireless Internet connection
Actually now I am getting so many other weird problems. First of all, choppy sound on the receiving end on the test server. I don't understand why all of a sudden voice will go choppy, when bandwidth and Internet upload and download speeds are good. On the production server, it registers but won't work. Sometimes when dialed, caller will still be listening the ringing tone even after the phone is picked up, and obviously no sound for the called party or caller because for caller still no one has picked up the phone. And when trying to call out, server will return error code 488. I don't know why but on the test server it all works fine, except for the choppy sound. Production server is exactly the copy of test server with all the same settings, but nothing works. I checked every single thing I could, but all the settings are the same. Maybe I am still missing some small little step. Its a GXP-2000 phones. Don't know what to do. On port 5060 all goes back to normal, but won't work if connected to Rogers Wireless Internet. On 5/14/07, Gerald A [EMAIL PROTECTED] wrote: Hi Zeeshan, On 5/13/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I've solved this problem. It was very easy (only if I knew how to do it before). I changed the UDP ports, i.e. 1. In sip.conf, bindport=5070 2. In my IP Phone server settings, www.myserver.com:5070 Now it seems to be working good and I hope there'll be no more problem with it. Sorry for not replying earlier; I got your note late, and then when I woke up had no Internet. Ah, the joys of Rogers. I'm glad to hear you solved it -- my only concern would be if you now want to connect ordinary 5060 looking phones. I will do a bit of research, I'm sure Asterisk can bind to more then one port. Thanks, Gerald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT (semi) E60 problem
On 14 May 2007, at 17:50, Martin Joseph wrote: Hello again gurus. I have been using Asterisk with great results going on a couple of years now. My primary box is running asterisk 1.42 built from a tar ball on Mac OSX 10.4.9. I have a very odd issue that I cannot seem to nail down, which is related to my Nokia E60 SIP phone. I use the E60 with very good results (latest firmware) from several locations. Basically it works fine from everywhere EXCEPT when it's on the same LAN as my asterisk box. The SIP config. that I have setup for the LAN connection refers to my asterisk box by it's local IP (ie 192.168.1.101). The external configs refer to the asterisk box by it's name (ie sip.domain.com). It seems like this has something to do with the authentication realm? If I create a new config on the phone using the LAN address, it works, but then when I leave the LAN, it appears to register, but issues a connection error when I try to place calls. To get it working again from outside the LAN, I can change the realm parameter in SIP.conf, and then reload and then change it AGAIN back to it's original value and reload, the phone then works fine. At that point, the LAN based config won't work anymore and will give me a connection error. Any thoughts on this? Ideas on how to troubleshoot further or work around it would be great. I had a similar problem, and it turned out to be capitalization error. The font the nokia uses for it's config dialogs is such that the caps don't stand out. This was compounded by the fact that the default input method capitalizes the first letter of every line. I guess this isn't your problem, but if it is I've saved you a _long_ tedious hunt! Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Some problems with mysql CDR
Hello,
We have finally upgraded to Asterisk 1.4, however we've run into two issues
that weren't occurring before the upgrade.
Issue #1: We're an outgoing call center and need to record all calls. We use
the uniqueid field in the CDR to match with the recording, which we labeled
with {UNIQUEID} in MixMonitor. For some reason, the uniqueid is not correct
in the CDR. Here is the manager event for a call:
Event: Cdr
Privilege: call,all
AccountCode: 6384106:MMI-Y:200705081051010077
Source: 00
Destination: 6398714109927773
DestinationContext: outbound
CallerID: 00
Channel: Zap/15-1
DestinationChannel: SIP/teliax-081ed5b0
LastApplication: NoOp
LastData:
StartTime: 2007-05-08 10:51:04
AnswerTime: 2007-05-08 10:51:05
EndTime: 2007-05-08 11:01:56
Duration: 652
BillableSeconds: 651
Disposition: ANSWERED
AMAFlags: DOCUMENTATION
UniqueID: 1178635864.1510
UserField:
And for that record in the database:
'calldate' '2007-05-08 10:51:04'
'clid' '00'
'src' '00'
'dst' '6398714109927773'
'dcontext' 'outbound'
'channel' 'Zap/15-1'
'dstchannel' 'SIP/teliax-081ed5b0'
'lastapp' 'NoOp'
'lastdata' '',
'duration' 652,
'billsec' 651,
'disposition' 'ANSWERED',
'amaflags' 3,
'accountcode' '6384106:MMI-Y:200705081051010077',
'uniqueid' '51010077',
'userfield' '',
'MMI_field' 'not found'
Issue #2: When a call is not answered, a record of that call is written to the
database, but uniqueid is left blank. The next time a call isn't answered,
Asterisk complains:
cdr_addon_mysql.c: mysql_cdr: Failed to insert into database: (1062) Duplicate
entry '' for key 1
I haven't found any other information regarding these errors. I am just
wondering if they are bugs. Any insight would be appreciated!
--
Jason Martin
Metrix Matrix, Inc.
785 Elmgrove Road, Building 1, Rochester, NY 14624
Office: 888-865-0065 Ext. 202
Mobile: (585) 721-8679
___
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[asterisk-users] Junghanns DuoBRI Card HELP !
I really need help. We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We keep on getting the error layer 1 deactivated (F3)! We then contacted Junghanns, who told us that there is no driver for the card for 1.4 and that we should try 1.2. We tried 1.2 with their driver but ALAS we get even more error messages! I don't understand how Junghanns puts an unsupported card on the market, but if anyone has any ideas, I would appreciate it. This card is long overdue for a client of ours. Apostolos ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Junghanns DuoBRI Card HELP !
I really need help. We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We keep on getting the error layer 1 deactivated (F3)! We then contacted Junghanns, who told us that there is no driver for the card for 1.4 and that we should try 1.2. We tried 1.2 with their driver but ALAS we get even more error messages! I don't understand how Junghanns puts an unsupported card on the market, but if anyone has any ideas, I would appreciate it. This card is long overdue for a client of ours. Apostolos ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ast_yyerror - Help
Hey all, We're starting to see all circuits are busy and a few dropped calls. When these happen, in the messages log, I see the following error. May 14 14:42:13 WARNING[5604] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: 0?7: What causes this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Junghanns DuoBRI Card HELP !
We have a Junghanns DuoBRI and we plugged it into an Asterisk 1.4 box. We keep on getting the error layer 1 deactivated (F3)! The card sees no ISDN device connected to it, neither in NT or TE modes alike. We then contacted Junghanns, who told us that there is no driver for the card for 1.4 and that we should try 1.2. We tried 1.2 with their driver but ALAS we get even more error messages! I don't understand how Junghanns puts an unsupported card on the market, but if anyone has any ideas, I would appreciate it. This card is long overdue for a client of ours. Apostolos ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk High-Capacity Stability
Thanks for all the input guys. This is what I had originally expected. Does anyone have any recommendations for other software configurations? I've thought about using OpenSER + rtpproxy(or media proxy), but it seems that OpenSER is not designed to do this sort of thing and would require some tricky hacking(?). I guess I'm wondering if their are any other opensource B2BUA-like softswitches that would fit what I'm looking for. What are these VoIP carriers using? Thanks, kn0x On 5/14/07, Daryl Jurbala [EMAIL PROTECTED] wrote: On May 14, 2007, at 1:29 PM, Zoa wrote: Several people do use it for handling 50k minutes a day. (I'm one of them). Yes, you need to know what you are doing, and have a nice design, but it is possible.Our code is only slightly altered. (mainly for billing purposes). That's great if you're good enough/have the time to make that happen. But when I have issues and call/pay Digium and don't get timely or meaningful answers, it's doesn't make for a good business decision to continue using it for that purpose when I can toss in a Nextone or Sansay and have it just work. All the time. No babysitting. Full professional and timely problem resolution from the vendor, etc, etc, etc. Don't even get me started on Digium not being able to get TC400Bs to properly negotiate g.723.1 5.3k when a client requests 6.3k first (thank god for Cantata). I guess it all comes down to whether you want things to just work and be able to have tier 1/2 support capable of actually doing anything meaningful, or if you want to have the engineering level people forced to do all the work. From my standpoint, the smart business decision is quite clear. But, as I said, Asterisk is still driving the feature servers, and works well for it. As mentioned by someone else previously in the thread, it makes a great endpoint. If you're having good success with it, that's fantastic. I would hope that you contribute back to the list how you set things up to make this a possibility. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue_exec: Unable to join queue
Is the queue enidan configured at all in queues.conf? and how is it defined? l. In data Mon, 14 May 2007 13:56:25 +0200, Per Jessen [EMAIL PROTECTED] ha scritto: I have a queue defined which I use like this: exten = _X.(reception),n,Ringing exten = _X.,n,Queue(enidan,t,,,20) exten = _X.,n,Voicemail(443,u) exten = _X.,n,Hangup() When I start asterisk, this queue doesn't work - -- Executing [EMAIL PROTECTED]:3] Queue(mISDN/3-u0, enidan|t|||20) in new stack [May 14 13:53:59] WARNING[17860]: app_queue.c:3541 queue_exec: Unable to join queue 'enidan' -- Executing [EMAIL PROTECTED]:4] VoiceMail(mISDN/3-u0, 443|u) in new stack But all I need to do to fix it is reload app_queue. Does anyone know what's going on? /Per Jessen, Zürich -- Home of QueueMetrics - http://queuemetrics.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel huge irq problem
You didn't even read the thread before replying. And for what it is worth, we at Digium are very anxious to solve any sort of IRQ problems that you (or others) might have. Matthew Fredrickson On May 14, 2007, at 1:43 PM, Salvatore Giudice wrote: Try switching to a Sangoma card. You won’t have anymore IRQ issues once you abandon Digium hardware. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Monday, May 14, 2007 12:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] zaptel huge irq problem Thanks Michael, I've already been through all that unfortunately, and I have a SATA drive, so no UDMA mode 2 as far as I know. I'm currently trying everything again anyway, but i doubt it will work if nothing worked the first time. Anyone would know of issues with XEN or SMP (or both) kernel? Do dual core AMD64 processors have issues? François. Michael L. Young wrote: François, I too had a similar problem and found the information on this page helpful: http://www.voip-info.org/wiki/view/Asterisk+PCI+bus+Troubleshooting What ended up working for me was changing the UDMA to mode 2 for the hard drive. Once I did that, this card has worked perfectly for me. Michael L. Young -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of François Delawarde Sent: Monday, May 14, 2007 10:24 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] zaptel huge irq problem Hello, I had noticed strange crackling sound on my phone calls going through my zaptel device (TDM400P), so i decided to check on possible timer issue, and found lots of issues on forums concerning the sensibility of zaptel with IRQs, and tried about everything: moving PCI slots, noapic and acpi=off boot options, play with different kernel options: iosched/preemption/timer/..., play with BIOS PCI options, change priorities, PCI latencies, IRQ balance, smp_afinity, but impossible to come up with anything correcting that problem. Any idea about this? Is it possible to force the timer to ztdummy (RTC timer) when you have a zap card plugged in? It's the only thing i could try to make it work. Thanks, François. Just in case: - Linux 2.6.18 with debian patches and xen enabled, asterisk running on dom0. - Here is my zttest results under a bit of load: # ./zttest Opened pseudo zap interface, measuring accuracy... 99.609375% 99.609375% 99.218750% 99.316406% 99.804688% 99.414062% 99.121094% 99.511719% 99.121094% 99.316406% 99.707031% 99.707031% 98.730469% 99.414062% 99.902344% 99.218750% 100.00% 99.414062% 98.828125% 99.218750% 99.316406% 98.449707% 100.00% - The card DOES NOT seem to share interrupts (checked also with lspci): # cat /proc/interrupts CPU0 CPU1 1: 1626 0 Phys-irq i8042 6: 3 0 Phys-irq floppy 8: 0 0 Phys-irq rtc 9: 0 0 Phys-irq acpi 14: 63 0 Phys-irq ide0 16: 1 0 Phys-irq libata, eth3 17: 6762583 0 Phys-irq libata 18: 13789 0 Phys-irq libata 19: 33459690 0 Phys-irq eth1 20: 19864325 0 Phys-irq sky2, eth0 21: 269250881 0 Phys-irq wctdm 256: 77735119 0 Dynamic-irq timer0 257: 3986325 0 Dynamic-irq resched0 258: 37 0 Dynamic-irq callfunc0 259: 0 4652748 Dynamic-irq resched1 260: 0 139 Dynamic-irq callfunc1 261: 0 28924306 Dynamic-irq timer1 262: 1021 0 Dynamic-irq xenbus 263: 0 0 Dynamic-irq console NMI: 0 0 LOC: 0 0 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ François Delawarde Ingeniero de red Tel: 918.03.92.51 E-mail: [EMAIL PROTECTED] _ WIRELESS MUNDI http://www.wirelessmundi.com/ C/Isaac Newton, 1 - Oficina 26 · Parque Tecnológico de Madrid 28760 TRES CANTOS (Madrid) Tlf./Fax: (+34) 918 03 92 51 La
Re: [asterisk-users] Asterisk High-Capacity Stability
Actually, OpenSER is just the you will need to scale Asterisk. We have perform a number of OpenSER to Asterisk implementation for 50k plus users -E On 5/14/07, Atlanticnynex [EMAIL PROTECTED] wrote: Thanks for all the input guys. This is what I had originally expected. Does anyone have any recommendations for other software configurations? I've thought about using OpenSER + rtpproxy(or media proxy), but it seems that OpenSER is not designed to do this sort of thing and would require some tricky hacking(?). I guess I'm wondering if their are any other opensource B2BUA-like softswitches that would fit what I'm looking for. What are these VoIP carriers using? Thanks, kn0x -- Thanks in advance and best regards, Ed Pimentel AgileCO Founder Web: http://AgileCO.net Mail: edpimentl[at]gmail.com Mail2: edpimentl[at]ieee.org IM: edpimentl [AOL | Jabber | Yahoo | MSN ] Voip: edpimentl [SKype | GoogleTalk ] Mobile Content Marketing/Management/Digital Delivery http://mobilecentral.ws Mobile ( Context Aware, AmbientIntelligence, Location ) based Social Network http://TagR.mobi (Alpha) Mobile Payment - P2P Payment http://agilepay.ws [S4]Secure Scalable Streaming Storage GridService http://DatR.ws Private Label Social Networks http://GooGaYa.com Sponsor of P2PSIP open source [viasip_ng] project Based on IETF P2PSIP WG https://sourceforge.net/projects/viasip/ http://groups.google.com/group/viasip_ng ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The purpose of DUNDi
remco, et al, could I use dundi where I could use an area code to determine the connecting server or dial string? just like we would use 88XXX to dial a 3 digit extension on another server at location 88? or dial 84XXX for a 3 digit extension on a server located at 84?... thanks, daveC Remco Post wrote: Rilawich Ango wrote: It is quite interesting and I am looking for it. Could you give me some more information or website how to set it up? Have a look at: http://atlaug.com/stuff/Presentations/Astricon06/JR_Richardson_Whitepaper.pdf and the two links at: http://www.voip-info.org/wiki/index.php?page=DUNDi%20Enterprise%20Configuration -- Building Strong Relationships w/ Intelligent Customer Service -- Interlocking Business Solutions, LLC 856-380-0894 x5000 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Proper AGI use with MySQL
Hi, We have a simple AGI script that provides some IVR functionality. It connects to a MySQL database in order to create a call record and capture IVR data. During the IVR process, we need to store the time the call started, so basically we INSERT a new MySQL row with callstart = NOW(), uniqueid = AGI(agi_uniqueid). As the user selects different options, we update the row to reflect the user's selection. There are a couple of options within the IVR that allows the user to speak with a live customer service rep. So, in those cases, we do a AGI exec to Dial out to the customer service queue and transfer the caller there. In the dialplan, we have extension h, execute DeadAGI which basically looks up the agi_uniqueid and updates the time the call ended in MySQL (e.g. callend = NOW()). All this seems to be working. However, we just don't feel we are doing things properly and reading up on the wiki more about AGI and dialing out, etc, just makes me feel we could be doing things better. Here are some of the things we think we could be doing better but are not sure: 1) Ideally, we would like for the AGI script to know when the call hangs up so that it properly updates callend without having to run the DeadAGI command in the h extension. 2) We would like for the AGI script to stay running for the life of the call and keep in memory all the user's IVR selections until the call is hung up. At which point, we could actually INSERT the row in MySQL with all the data, instead of constantly hitting the database with updates. 3) We read on the wiki the following: If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. In our IVR, we always exit with -1. So, this statement confused us. Does it mean that when we transfer the call to the queue, we should actually return 0 instead of -1 to indicate that the AGI is still running? Can anyone explain this further? 4) When should we close the database handle? Currently, we have it at the end of the AGI script and also as part of the DeadAGI script. However, which one is actually closing it, we don't know. Comments are extremely welcomed and appreciated. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Proper AGI use with MySQL
Sorry, just to make sure this is clear, in #2 below, when I said We would like for the AGI script to stay running for the life of the call..., I also meant after the call is transfered to the customer service queue. This is so because we need to note that the call ended (update callend = NOW()) regardless of whether the call stayed only in the IVR or the caller spoke with a customer service agent. Thanks again On Mon, May 14, 2007 5:40 pm, [EMAIL PROTECTED] said: Hi, We have a simple AGI script that provides some IVR functionality. It connects to a MySQL database in order to create a call record and capture IVR data. During the IVR process, we need to store the time the call started, so basically we INSERT a new MySQL row with callstart = NOW(), uniqueid = AGI(agi_uniqueid). As the user selects different options, we update the row to reflect the user's selection. There are a couple of options within the IVR that allows the user to speak with a live customer service rep. So, in those cases, we do a AGI exec to Dial out to the customer service queue and transfer the caller there. In the dialplan, we have extension h, execute DeadAGI which basically looks up the agi_uniqueid and updates the time the call ended in MySQL (e.g. callend = NOW()). All this seems to be working. However, we just don't feel we are doing things properly and reading up on the wiki more about AGI and dialing out, etc, just makes me feel we could be doing things better. Here are some of the things we think we could be doing better but are not sure: 1) Ideally, we would like for the AGI script to know when the call hangs up so that it properly updates callend without having to run the DeadAGI command in the h extension. 2) We would like for the AGI script to stay running for the life of the call and keep in memory all the user's IVR selections until the call is hung up. At which point, we could actually INSERT the row in MySQL with all the data, instead of constantly hitting the database with updates. 3) We read on the wiki the following: If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. In our IVR, we always exit with -1. So, this statement confused us. Does it mean that when we transfer the call to the queue, we should actually return 0 instead of -1 to indicate that the AGI is still running? Can anyone explain this further? 4) When should we close the database handle? Currently, we have it at the end of the AGI script and also as part of the DeadAGI script. However, which one is actually closing it, we don't know. Comments are extremely welcomed and appreciated. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: dialplan: execute on hangup
In article [EMAIL PROTECTED], Michael Kamleitner [EMAIL PROTECTED] wrote: thx Tony, but DeadAGI doesn't seem to fit my needs... the way I understand its functioniality ( http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI), DeadAGI is ensureing that an executed AGI-script is finished, even if the caller hung up _during_ execution. This is true, but DeadAGI will also work when the channel is already hung up. in my case, I need to execute the AGI-script _after_ the user hung up the voicemail is recorded. That's exactly what DeadAGI is for. I use DeadAGI extensively within the 'h' extension to do post-call processing for my applications. another ideas: is there away to tell the Voicemail-command to execute an AGI-script when recording is finsihed? Not without modifying the source code. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: dialplan: execute on hangup
thx a lot Tony, I didn't know about using the h-extension (I'm new to
Asterisk)!
this way it works:
...
exten = s,n,Voicemail(${Enter},u)
exten = s,n,AGI(foneboxx.php|${Enter})
exten = h,1,DeadAGI(foneboxx.php|${Enter})
greetings,
michael
On 5/14/07, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],
Michael Kamleitner [EMAIL PROTECTED] wrote:
thx Tony, but DeadAGI doesn't seem to fit my needs... the way I
understand
its functioniality (
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+DeadAGI),
DeadAGI
is ensureing that an executed AGI-script is finished, even if the caller
hung up _during_ execution.
This is true, but DeadAGI will also work when the channel is already hung
up.
in my case, I need to execute the AGI-script _after_ the user hung up
the
voicemail is recorded.
That's exactly what DeadAGI is for. I use DeadAGI extensively within the
'h'
extension to do post-call processing for my applications.
another ideas: is there away to tell the Voicemail-command to execute an
AGI-script when recording is finsihed?
Not without modifying the source code.
Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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--
Mag. Michael Kamleitner
-
[EMAIL PROTECTED]
https://www.xing.com/profile/Michael_Kamleitner
-
m-otion GmbH
Favoritenstr 4-6/III, 1040 Wien
+43 1 205705 / 21 (Fax 99)
-
www.m-otion.com
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Re: [asterisk-users] ast_yyerror - Help
On Mon, 2007-05-14 at 14:52 -0500, Rob Schall wrote:
Hey all,
We're starting to see all circuits are busy and a few dropped calls.
When these happen, in the messages log, I see the following error.
May 14 14:42:13 WARNING[5604] ast_expr2.fl: ast_yyerror(): syntax error:
syntax error, unexpected $end, expecting TOK_MINUS or TOK_COMPL or
TOK_LP or TOKEN; Input:
0?7:
What causes this?
The ast_expr2 stuff is what gets called when you have $[...] expressions
in your dialplan.
So, it looks like you have have something like this in your dialplan:
$[ ${var} ? ${var2} : ${var3} ]
and ${var3} evaluates to an empty string (or something).
You need to narrow down where exactly in the dialplan you are when this
error happens. If you have no idea, look for $[ in your dialplan, and
study each one to determine a set of candidates. Once you find the right
expression, then you need to determine why var3 is empty, and maybe
insert some code to make sure it's always set to something, or rephrase
the expression to work better in that case.
Best of luck!
murf
--
Steve Murphy
Software Developer
Digium
smime.p7s
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[asterisk-users] Asterisk Now
Can someone tell me what is included in this distro? Does it have voicemail, meetme, panel, and IVR? Thanks, Wiley E. Siler Director of Information Technology 4110 N. Scottsdale Rd. Ste 110 Scottsdale, Arizona 85251 (480) 296.0260 (866) 320.2083 ext. 1003 mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.education2020.com http://www.education2020.com/ Helping students on a mission. Graduation and beyond. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Blind Transfer - Who transferred the call?
Hi all,
Is there a way to tell which extension transferred a call in a blind
transfer?
Sorry if it's a basic question, but I haven't seen an answer.
${CALLERID(num)} still holds the outside party caller id (which it
should), but I'd like to the extension number of the extension that
transferred the call.
Any suggestions?
Thank you,
--
Warm Regards,
Lee
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RE: [asterisk-users] Proper AGI use with MySQL
How about forking the process when the AGI launches, and pass the PID back to Asterisk in a variable. When the call ends (caught at the h), call another AGI script to kill/stop that pid. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, May 14, 2007 5:45 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Discussion Subject: RE: [asterisk-users] Proper AGI use with MySQL Sorry, just to make sure this is clear, in #2 below, when I said We would like for the AGI script to stay running for the life of the call..., I also meant after the call is transfered to the customer service queue. This is so because we need to note that the call ended (update callend = NOW()) regardless of whether the call stayed only in the IVR or the caller spoke with a customer service agent. Thanks again On Mon, May 14, 2007 5:40 pm, [EMAIL PROTECTED] said: Hi, We have a simple AGI script that provides some IVR functionality. It connects to a MySQL database in order to create a call record and capture IVR data. During the IVR process, we need to store the time the call started, so basically we INSERT a new MySQL row with callstart = NOW(), uniqueid = AGI(agi_uniqueid). As the user selects different options, we update the row to reflect the user's selection. There are a couple of options within the IVR that allows the user to speak with a live customer service rep. So, in those cases, we do a AGI exec to Dial out to the customer service queue and transfer the caller there. In the dialplan, we have extension h, execute DeadAGI which basically looks up the agi_uniqueid and updates the time the call ended in MySQL (e.g. callend = NOW()). All this seems to be working. However, we just don't feel we are doing things properly and reading up on the wiki more about AGI and dialing out, etc, just makes me feel we could be doing things better. Here are some of the things we think we could be doing better but are not sure: 1) Ideally, we would like for the AGI script to know when the call hangs up so that it properly updates callend without having to run the DeadAGI command in the h extension. 2) We would like for the AGI script to stay running for the life of the call and keep in memory all the user's IVR selections until the call is hung up. At which point, we could actually INSERT the row in MySQL with all the data, instead of constantly hitting the database with updates. 3) We read on the wiki the following: If the AGI application dials outward by executing Dial, control over the call returns to the dialplan and the script loses contact with the Asterisk server. The script continues to run in the background by itself and is free to clean up and do post-dial processing. In our IVR, we always exit with -1. So, this statement confused us. Does it mean that when we transfer the call to the queue, we should actually return 0 instead of -1 to indicate that the AGI is still running? Can anyone explain this further? 4) When should we close the database handle? Currently, we have it at the end of the AGI script and also as part of the DeadAGI script. However, which one is actually closing it, we don't know. Comments are extremely welcomed and appreciated. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
