Re: [asterisk-users] s-${DIALSTATUS} extensions
Make it
Goto(s-${DIALSTATUS})
cheerz
- Ben.
Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial
attempts in the s extension. Goto() is used in examples. Is the
prefix s- mandatory? Is it related to the original extension s?
(Apparently Goto(${DIALSTATUS}) won't work for me.)
Yuan Liu
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Re: [asterisk-users] s-${DIALSTATUS} extensions
On Tue, Feb 06, 2007 at 11:58:01PM -0800, Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial attempts
in the s extension. Goto() is used in examples. Is the prefix s-
mandatory? Is it related to the original extension s? (Apparently
Goto(${DIALSTATUS}) won't work for me.)
No. the text 's-' is an arbitrary text in this case.
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[asterisk-users] Type of wake-up Call
Hi there, Is there a way to program asterisk to dial an extension Monday to Friday at a specific time and then read a specific string? eg: Kids, go to the bus stop now, you're about to miss the bus! Many thanks, Pierre ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] s-${DIALSTATUS} extensions
On Tue, 6 Feb 2007, Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial attempts in
the s extension. Goto() is used in examples. Is the prefix s- mandatory?
Is it related to the original extension s? (Apparently Goto(${DIALSTATUS})
won't work for me.)
s- is just something tagged onto the label. An example of what I use:
exten = s,n,Dial(${ARG1},${timeOut},ron)
exten = s,n,Noop(Initial dial failed: ${DIALSTATUS})
exten = s,n,Goto(${DIALSTATUS})
exten = s,n(ANSWER),Noop(Answered)
exten = s,n,Hangup()
exten = s,n(NOANSWER),Noop(Starting NOANSWER processing)
And so on...
the extension 's' in those lines above is the extension inside a macro.
Gordon
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[asterisk-users] Pickup
On one installation (* 1.2.13) Pickup doesn't work. This is what I have in
extensions.conf
exten = _**2X,1,Pickup(${EXTEN:2}8${EXTEN:3}tuevents)
exten = _**2X,n,Hangup
This is what I get on CLI
-- Executing NoOp(mISDN/3-1, incoming-beronet 80 - dolazni poziv s broja
270248) in new stack
-- Executing LookupCIDName(mISDN/3-1, ) in new stack
-- Executing Dial(mISDN/3-1, SIP/20|30|t) in new stack
-- Called 20
-- SIP/20-08cdad80 is ringing
Extension Changed 20 new state Ringing for Notify User 27
Extension Changed 20 new state Ringing for Notify User 21
Extension Changed 20 new state Ringing for Notify User 28
-- Incoming call: Got SIP response 415 Unacceptable Content-Type back from
192.168.2.107
Extension Changed 27 new state InUse for Notify User 21
Extension Changed 27 new state InUse for Notify User 20
Extension Changed 27 new state InUse for Notify User 28
-- Executing Pickup(SIP/27-b65a1100, 2080tuevents) in new stack
== Spawn extension (sip2, **20, 1) exited non-zero on 'SIP/27-b65a1100'
Extension Changed 27 new state Idle for Notify User 21
Extension Changed 27 new state Idle for Notify User 20
Extension Changed 27 new state Idle for Notify User 28
Why do I get == Spawn extension (sip2, **20, 1) exited non-zero on
'SIP/27-b65a1100'
I have to pickup either 2X, 8X, t or uevents extension (phone will ring on any
of those).
Have I done something wrong?
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Re: [asterisk-users] Re: Help - Poor Voice Quality
On 7 Feb 2007, at 03:59, Jim Duda wrote: Thanks for the reply Lacy. Yes, I know that I am using IAX2 and not SIP for my connection to teliax. IAX2 is the preferred protocol for connection to teliax. I have the firewall configured to prioritorize port 4569 for IAX2. I have the shorewall tcdevices file setup with 3 mbit download and 500 Kbit upload speeds. We need a few clues :-) What sort of 'poor' quality are we talking about - when folks complain what words do they use? Which codec(s) are you using? How many channels do you want to use at once ? What is the round-trip time between you and the teliax server ? Do you have the jitterbuffer on or off ? (if you only have 6ms of jitter, I'd switch it off) 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] registration not timing out?
every few days my ADSL connection gets dropped for a few seconds. When it does I find my SIP connection to one of my providers does not timeout and retry. Does the following give some clues? Asterisk 1.2.13, Copyright (C) 1999 - 2006 Digium, Inc. and others. (note this is the debian etch/testing package, I can build a new one if needed) .. CLI sip show registry HostUsername Refresh State iinettrunk:5060 [EMAIL PROTECTED] 3584 Request Sent sip.pennytel.com:5060 N 280 Registered When I see this I can do this to fix it up (note the error message). CLI sip reload Reloading SIP Feb 7 20:07:51 NOTICE[2889]: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 4208! == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found Now it's all good: sip show registry HostUsername Refresh State iinettrunk:5060 [EMAIL PROTECTED] 3584 Registered sip.pennytel.com:5060 88809289 280 Registered firewall3*CLI ___ --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] Disconnection supervision: what about PBX
At 05.23 07/02/2007, you wrote: Yuan LIU wrote: After reading through several recent threads, I started to wonder why the Cisco document (and other VoIP documents) appears to present this issue as VoIP gateway specific. Don't (plain old) PBX' face the same issue if they use analogue interfaces? If there are analogue PBX' at all, how do they solve the problem? Yes, analog PBXs have the same issues. Don't do anything to solve the issue. That is way many hotels tell their guests to not let a call ring for more than 45 seconds or the call will be billed even if it was not answered. I agree with LIU. A standard analog PBX tries to solve these billing problems (for example in Italy you have a billing pulse from the telco that can be intercepted by analog PBX and thus billed). Why shouldn't Asterisk try to do the same? There's too much confusion about call progress functionality, in Asterisk code and documentation. Shouldn't be better to say EITHER that it can work in any country but there's still too much work to do OR that it cannot work and then take it away from the source code? I mean if there's a way to make it work (using different systems for different countries), then I think it's an important feature (considering also that many companies including Digium sell FXO module for analog lines). If there is no way, better maybe just get rid of it and put a red sign on the product specifications of the analg cards YOU'LL NOT BE ABLE TO DO BILLING!!!. Rgds. Stefano ___ --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] Type of wake-up Call
Hi, Am 07.02.2007 um 09:53 schrieb Pierre du Plessis: Is there a way to program asterisk to dial an extension Monday to Friday at a specific time and then read a specific string? eg: Kids, go to the bus stop now, you're about to miss the bus! Write a cronjob which creates a call file. Shouldn't be a big thing. In case you are not familiar with call files: Create a file dummy.call with the following content. ---cut--- Channel: SIP/2000 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: call-file-test Extension: 10 ---cut SIP/2000 being the phone on your desk. And add the following context to your dialplan: ---cut--- [call-file-test] exten = 10,1,Answer() exten = 10,n,Wait(1) exten = 10,n,Playback(hello-world) exten = 10,n,Wait(1) exten = 10,n,Hangup() ---cut--- Move the dummy.call file to /var/spool/asterisk/outgoing/ and wait. PS. You can touch a call file to be executed in the future. But I'd prefer the cronjob. Stefan -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] one touch recording problem in asterisk 1.4
Hi. I was using asterisk 1.2 on a box with sip phones attached and a long distance T1 line as the phone provider. We did a successful test of *1 allowing one-touch recording as set in the features.conf. Because of deadlock issues I decided to try 1.4 (latest svn as of yesterday) and the deadlock went away, but when we tried to use the *1 it was sent over the bridged channel rather than being responded to by the local box. I just see the start and end of vldtmf by the zap chnnel driver and that is it. I made no other configuration changes, so am I doing something wrong or is there a bug? Thanks in advance for all your help. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici [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] dnsmgr seems to have died
Hello, A few weeks ago I enabled the dnsmgr. A few days ago I noticed we could not reach any IAX2 peers in the USA. I did everything I could think of including a full reboot to no avail. Re-commenting the enable in dnsmgr.conf and restarting asterisk made things work again. Have there been other reports about this? (I saw none in a search of the list). This is with asterisk 1.2 which has been running perfectly since 1.2 was released. ___ --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] Having Trouble With Wait Command in Callback Context
exten =h,2,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) You could run a script instead of the cp command in system and add the wait in that. ___ --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] Glitches in voicemail prompts
I changed from using a recent asterisk system standalone to a Trixbox install and now I get clicks and minor dropouts on the voicemail prompts. System load is non-existant on this machine, interrupts *appear* to be fine, and as near as I can tell the glitch is at the same point in the prompt each time... Any suggestions on how to debug this further? To my ear it sounds like what happens when you get an overflow in some decoder code and the levels have wrapped around? Any thoughts? Ed W ___ --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] Disconnection supervision: what about PBX
From: Yuan LIU Sent: Tuesday, February 06, 2007 8:11 PM After reading through several recent threads, I started to wonder why the Cisco document (and other VoIP documents) appears to present this issue as VoIP gateway specific. Don't (plain old) PBX' face the same issue if they use analogue interfaces? If there are analogue PBX' at all, how do they solve the problem? Yuan, Well engineered analogue PBXs typically do not use standard loop start subscriber lines. When digital trunks are not an option, they use analogue PBX and/or DID trunks. At the very least, ground start circuits are preferred to avoid glare. The best call quality for analogue is achieved by using four-wire EM trunks that provide answer and disconnect supervision. There are two-wire trunks (which are probably more common), as well as different signalling methods. These trunks require special interface hardware, and I am unaware of any that work with Asterisk. As the cards are typically very expensive, it is usually better to go with digital if you require that functionality. It would be nice to see a BRI interface for Asterisk that works in North America, as BRI circuits are often comparable in price to analogue lines. Sincerely, Trevor Hammonds ___ --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] Are there any IP phone in the market have suchfeatures?
Many SIP phones can use the SUBSCRIBE/NOTIFY mechanism of RFC-3265 to subscribe to hints in Asterisk. This can be used to show e.g. parking bay and/or agent status. Have a look at http://www.voip-info.org/wiki/view/Asterisk+standard+extensions . -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Xue Liangliang Sent: 07 February 2007 02:31 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Are there any IP phone in the market have suchfeatures? Hi, all, Do you guys happen to know that there are any IP phones have such feature, that it can has some indication for the agent status linked to the phone? E.g some LED show the status, backend we can link the phone to one agent id, then the agent login the system, the 'online' indication will be blinking and on, if logout with type of meeting, then 'meeting' LED will be on, and etc for other scenarios. I found it is quite common in the traditional PABX, however now with more advanced technology, we lost such features. Regards, Liangliang ___ --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] SIP/Console - ISDN ticks
I am experiencing audio ticks when doing calls from SIP or console to ISDN. Calls. Everything appears fine when doing ISDN-ISDN or SIP-SIP. Console calls results in 5-8 ticks a second, SIP calls are dependent on buffer size - 16ms are 1 tick a second, 8ms are 2-3 ticks a second. I recently moved the ISDN board and software to an upgraded PC with the exact same software configuration - prior to moving the audio was fine. There are no IRQ conflicts, and just for the heck of it I have experimented with moving the card to another PCI slot, enabling/disabling APIC/ACPI in kernel, using version 1.4.0 and trunk software. I have also tried to load and unload the computer using hdparm and other testprograms, it does not affect the ticking or zttest results. zttool reports no IRQ misses. zttest results are normally 99.755 but occasionally (once 2-3 minutes) dips to 99.96 or worse (99.92). - using ztdummy for timing and then running zttest is all bad I have tried to adjust the clocks using adj_clock (in zaptel source) by Cohen Tzafrir (Not knowing exactly what it would improve), it did adjust the clock to 100% with an occasional dip (every 2-3 minutes as before) - but it didn't affect audio ticks, which is always consistently bad. Zaptel is not compiled using MMX support. I would really appreciate any thoughts on this !!! Regards Rasmus ___ --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 Cmd to ID Mobile from Phone#?
Is there an Asterisk command, app, AGI (or other) that can be called with a phone# (or list) that will lookup somewhere definitive and report whether the phone# is registered to a mobile phone or not? How about other data, like its home city/district etc? -- (C) Matthew Rubenstein ___ --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 Cmd to ID Mobile from Phone#?
Take a look at smartCID (at www.generationd.com) Does a reverse lookup for name/location/etc. Based on phone number. MD -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: Wednesday, February 07, 2007 8:30 AM To: Asterisk-Users Subject: [asterisk-users] Asterisk Cmd to ID Mobile from Phone#? Is there an Asterisk command, app, AGI (or other) that can be called with a phone# (or list) that will lookup somewhere definitive and report whether the phone# is registered to a mobile phone or not? How about other data, like its home city/district etc? -- (C) Matthew Rubenstein ___ --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] H323 to SIP - One way voice
Hello all, I want to use asterisk as protocol converter, H323 to SIP. I am using Asterisk 1.2.14 with chan_h323 and the free version of g729. When calling from SIP to H323 everything is fine. But when calling from H323 to SIP, the phone using SIP doesn't hear the other party. The phones and Asterisk are in the same subnet and the firewall of the Asterisk box is off. Please advice. Thank you, Andrei U ___ --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: Mabe OT? What managed switch is best for VoIP application?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I worked with Cisco and HP and they should do what you are looking for. I even worked with cheap unmanaged switches ~20 Euro and they work with VoIP. Do you know for switch that can tell me that on port 7 there are two active SIP calls. One of them goes to x.x.x.x IP address and another to sip.mydomain.com. First lasts for 34 and another 51 seconds. -- Tomislav Parcina [EMAIL PROTECTED] winmail.dat___ --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 severe sound problems
Hi everyone! I'm using Asterisk 1.2.7.1 on a CentOS 4 server with 5 - 9 agents and I'm having some issues with the Chanspy application. All the agents are on SIP channels with g711 and all the communications are inside a LAN. When I'm spying a SIP channel, the audio from one of the ends (normally the caller) sounds *extremely* (unusable) choppy, as if it was losing some frames. Sometimes the called party is heard almost perfectly, but there are ALWAYS sound quality issues. The agents do not report any problem, and the audio recorded with the Monitor applications sounds reasonably fine. I'm able to reproduce the problem with any amount of load and it happened also while doing tests with my computer as an Asterisk server. Additional Information: * Asterisk 1.2.7.1 built by test @ ast3 on a i686 running Linux on 2006-04-24 10:52:49 UTC * Linux foo.bar.com 2.6.9-34.0.2.ELsmp #1 SMP Fri Jul 7 19:52:49 CDT 2006 i686 i686 i386 GNU/Linux * Intel(R) Pentium(R) 4 CPU 3.00GHz, 1GB RAM. did anyone encountered the same situation? Google only reported one similar problem without a solution (http://bugs.digium.com/print_bug_page.php?bug_id=7340) any ideas are welcome! thanks a lot! saludos, -- santiago aguiar *netlabs* / Palmar 2548 Montevideo, Uruguay Tel. +(598 2) 707 7687 Fax. +(598 2) 709 4866 / http://www.netlabs.com.uy begin:vcard fn:Santiago Aguiar n:Aguiar;Santiago org:;Desarrollo adr:;;Palmar 2548;Montevideo;Montevideo;11600;Uruguay email;internet:[EMAIL PROTECTED] title:NetLabs tel;work:+598 2 7077687 tel;fax:+598 2 7094866 tel;home:+598 2 7075079 tel;cell:+598 99 579739 x-mozilla-html:TRUE url:http://www.netlabs.com.uy/ version:2.1 end:vcard ___ --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] pridialplan/prilocaldialplan
Christoph Fürstaller schrieb: Can someone explain what the parameters pridialplan and prilocaldialplan are? What do they do and do I need them? I've connected an asterisk box via E1 (sangoma) to an alcatel 4200 pbx. The pbx technican complains about the format of the nr asterisk sends. Asterisk sends all numbers in on piece the pbx expects the numbers devided into international prefix, national prefix, phone number and extension. How can I set this behaviour? Is this possible with the above mentioned parameters? Or do I need something else/different? I hope someone can explain that to me. Hello ! pridialplan/prilocaldialplan sets the type of number information for the called/calling number. You need to be more specific which number you mean (called or calling) The pbx technican is wrong unless the alcatel uses a different protocol than EuroISDN or QSIG. Regards Hans ___ --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
R: [asterisk-users] pridialplan/prilocaldialplan
Look at here http://lists.digium.com/pipermail/asterisk-users/2005-May/102837.html Giordano -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Johann Steinwendtner Inviato: mercoledì 7 febbraio 2007 14.56 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [asterisk-users] pridialplan/prilocaldialplan Christoph Fürstaller schrieb: Can someone explain what the parameters pridialplan and prilocaldialplan are? What do they do and do I need them? I've connected an asterisk box via E1 (sangoma) to an alcatel 4200 pbx. The pbx technican complains about the format of the nr asterisk sends. Asterisk sends all numbers in on piece the pbx expects the numbers devided into international prefix, national prefix, phone number and extension. How can I set this behaviour? Is this possible with the above mentioned parameters? Or do I need something else/different? I hope someone can explain that to me. Hello ! pridialplan/prilocaldialplan sets the type of number information for the called/calling number. You need to be more specific which number you mean (called or calling) The pbx technican is wrong unless the alcatel uses a different protocol than EuroISDN or QSIG. Regards Hans ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.29/673 - Release Date: 06/02/2007 17.52 -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.29/673 - Release Date: 06/02/2007 17.52 ___ --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] Type of wake-up Call
Stefan Wintermeyer wrote: Hi, Am 07.02.2007 um 09:53 schrieb Pierre du Plessis: Is there a way to program asterisk to dial an extension Monday to Friday at a specific time and then read a specific string? eg: Kids, go to the bus stop now, you're about to miss the bus! Write a cronjob which creates a call file. Shouldn't be a big thing. In case you are not familiar with call files: Create a file dummy.call with the following content. ---cut--- Channel: SIP/2000 MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: call-file-test Extension: 10 ---cut SIP/2000 being the phone on your desk. And add the following context to your dialplan: ---cut--- [call-file-test] exten = 10,1,Answer() exten = 10,n,Wait(1) exten = 10,n,Playback(hello-world) exten = 10,n,Wait(1) exten = 10,n,Hangup() ---cut--- Move the dummy.call file to /var/spool/asterisk/outgoing/ and wait. PS. You can touch a call file to be executed in the future. But I'd prefer the cronjob. Stefan -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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 You could probably modify wakeup.php http://voip-info.org/wiki/view/Asterisk+tips+Wake-Up+Call+PHP signature.asc Description: OpenPGP digital signature ___ --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] Connection problem w/ Attended Transfer
Hi all, I'm new posting here, though not to perusing. I'm having an issue with attended transfer and was wondering if anyone had heard of the problem/had any suggestions... Apologies in advance if this post is excessively newb-oid. - An incoming call C is passed to A, a POTS telephone connected via a Handytone 286 ATA. - A presses atxfer key, then dials B, a Win XP laptop running x-lite. - A and B talk and A hangs up to transfer C to B. - Most audio between B and C is lost, for the small proportion that does get through, latency is very high. - When B and C hang up, asterisk sometimes 'crashes' - incoming calls are rejected and the CLI becomes unresponsive to commands. Asterisk version is 1.2.14. An example of the cli output with max verbosity is at http:// nyodrinkers.com/cliout.txt I know there have been problems with call transfers the Handytone line, I recently updated the firmware which fixed blind transfer and attended transfer at least now works in theory... If anyone can help I'd be massively grateful! Best wishes, Ben Hall extensions.conf: [voiptalkincoming] exten = 01225808102,1,Answer exten = 01225808102,2,Dial(SIP/reception,10,t) ; at this point 'reception' [ie A] dials 100 exten = 100,1,Dial(SIP/mrblobby,10,t) ; the quality of the transferred call between mrblobby and exten = 100,2,Hangup ; voiptalk [ie B and C] is extremely poor sip.conf [general] jbenable = yes jbmaxsize = 1000 jbresyncthreshold = 1000 [reception] type=friend user=reception secret= callerid=Ben host=dynamic nat=no [EMAIL PROTECTED] allow=all context=outgoing [mrblobby] type=friend user=mrblobby secret= callerid=Blobby host=dynamic nat=no [EMAIL PROTECTED] allow=all context=outgoing ___ --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] prob with not recognizing hangup, pickup - python
Hello there all. i have an agi-bin python script that calls out when a file is dropped into the /var/spool/outgoing the script seems to work, and the call is placed, but the script runs without knowing when the phone is picked up. i mean, the call is made, and the script begins to run. So by the time it is answered, most of it has already played out. is there a way that asterisk can initiate a phone call and wait to run the script untill after the phone is answered? also, is there a way to know when the line is disconnected ? I would like to write a function that would kill my script if the phone is disconnected. This is for an IVR application where we phone out to our customers if the status of one of their machines changes thanks for any tips, shawn ___ --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] New Issue
I'm still not seeing chan_zap in menu option three. I copied the source directories from /root/downloads/asterisk (where I had put them) to /usr/src/ and then did what you suggested below and I got the same result. I'm going to try make uninstalling all the packages deleted all source directories and starting over from the downloads. If you any other suggestions I'll do them. TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cosmin Prund Sent: Tuesday, February 06, 2007 6:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New Issue Try it like this: cd /usr/src/asterisk-1.4.0 make clean ./configure --with-zaptel=/usr/src/zaptel-1.4 make menuconfig make all make install David Ruggles wrote: Sorry about that I must have been in the wrong directory. I also have 1.4.0 and I tried it again and it worked. Chan_zap is not listed there, I'll start poking around and see if I stumble across anything. Do you know where the expected location is? I don't have a problem moving the source. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[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] Re: Cordless SIP Phones
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Siemens Gigaset IP phones (C450-IP, S450-IP) are not that bad (gigaset.siemens.com). C450IP costs less than 100 USD (in Italy at least), S450 is slightly more expensive. I have Siemens C450 IP for two days and it seams weary good. I'm looking for S450 IP, but I can't buy it in Croatia :( -- Tomislav Parcina [EMAIL PROTECTED] winmail.dat___ --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] New Issue
the ./configure thing requires the sources of zaptel, not asterisk. Are you sure they're passing the zaptel sources? Well... i'm out of ideas. If it doesn't work you might want to re-post your thread (specifically say you don't see chan_zap in make menuconfig) and start with a new message (send) - don't reply to an existing message and change it's subject line. When you first posted this message you hijacked a thread called Mysterious tables starting with stats_. People using threaded mail readers might not even see your question! I saw your question because the thread about Mysterious stats_ tables looked interesting... David Ruggles wrote: I'm still not seeing chan_zap in menu option three. I copied the source directories from /root/downloads/asterisk (where I had put them) to /usr/src/ and then did what you suggested below and I got the same result. I'm going to try make uninstalling all the packages deleted all source directories and starting over from the downloads. If you any other suggestions I'll do them. TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cosmin Prund Sent: Tuesday, February 06, 2007 6:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] New Issue Try it like this: cd /usr/src/asterisk-1.4.0 make clean ./configure --with-zaptel=/usr/src/zaptel-1.4 make menuconfig make all make install David Ruggles wrote: Sorry about that I must have been in the wrong directory. I also have 1.4.0 and I tried it again and it worked. Chan_zap is not listed there, I'll start poking around and see if I stumble across anything. Do you know where the expected location is? I don't have a problem moving the source. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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 ___ --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] Billing pulses
Hello, I've discovered that in Italy ISDN lines can be programmed to generate a billing pulse every n seconds (it dipends from the pricebook). The pulse has these figures: frequency 12 kHz ± 1% level .. 200 mVrms on 200 distortion... 5% pulse duration .125 ± 25 ms pause duration 180 ms period ... 300 ms Does someone know if these values can be used somehow to get an accurate billing using asterisk with these lines? Could be a matter of configuration or programming? Thanks Stefano ___ --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] s-${DIALSTATUS} extensions
Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial
attempts in the s extension. Goto() is used in examples. Is the prefix
s- mandatory? Is it related to the original extension s? (Apparently
Goto(${DIALSTATUS}) won't work for me.)
Goto(${DIALSTATUS}) won't work because with only one parameter Goto will
think it is a priority. Try Goto(${DIALSTATUS},1)
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Re: [asterisk-users] Billing pulses
Funny that a digital line have a analogue pulse. Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system. Jorge Mendoza Stefano Corsi wrote: Hello, I've discovered that in Italy ISDN lines can be programmed to generate a billing pulse every n seconds (it dipends from the pricebook). The pulse has these figures: frequency 12 kHz ± 1% level .. 200 mVrms on 200 distortion... 5% pulse duration .125 ± 25 ms pause duration 180 ms period ... 300 ms Does someone know if these values can be used somehow to get an accurate billing using asterisk with these lines? Could be a matter of configuration or programming? Thanks Stefano ___ --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] Billing pulses
At 16.22 07/02/2007, you wrote: Funny that a digital line have a analogue pulse. Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system. Yes, it's strange. But I find no mention on answer supervision in the NT1Plus manual (NT1Plus is the hardware device the Telco installs when you ask for an ISDN line). Where should I ask for answer supervision? The Telco? That sounds very difficult in Italy... they have no technical call centers. Almost only sales. But if the line should provide those analog billing pulses... do you think could be possible to intercept them? Rgds Stefano ___ --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 access environment variable?
'export MYIP' in the startup script for Asterisk.
Larry Alkoff wrote:
I was only trying to demonstrate that my special variable MYIP was
indeed in the environment of the shell. I suspect it's not in the
Asterisk process environment - why I dunno.
I'll look at that tomorrow but suspect I'll never be able to read the
MYIP variable from Asterisk.
Larry
Tzafrir Cohen wrote:
On Tue, Feb 06, 2007 at 08:04:23AM -0600, Larry Alkoff wrote:
Thanks for your reply Ioan.
Very interesting. ${ENV(PATH)} works to display the path
but ${ENV(MYIP)} does not!
There must be a list in Asterisk that only allows cerain
environmental variables to be shown. A very unnecessary bummer.
Right.
However, at the CLI prompt:
! echo $PATH and ! echo $MYIP
both work fine.
However This is incorrect: '!' only works in a remote asterisk
terminal: a connection from a different process (on the same system)
to the running
Asterisk process.
It will run a subshell of thatremote process. So it is not necessarily
related to the environment of the Asterisk process.
Also: when running something in System(), note that you run a
subprocess, and that this subprocess may have its own separate
environment.
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Re: [asterisk-users] Re: Help - Poor Voice Quality
Jim, I too am a Teliax user. Talk to their technical support. IAX2 is NOT preferred. They'll tell you to use SIP. Jim Duda wrote: Thanks for the reply Lacy. Yes, I know that I am using IAX2 and not SIP for my connection to teliax. IAX2 is the preferred protocol for connection to teliax. I have the firewall configured to prioritorize port 4569 for IAX2. I have the shorewall tcdevices file setup with 3 mbit download and 500 Kbit upload speeds. Jim Lacy Moore wrote: Jim Duda wrote: I've been on the shorewall firewall and confirmed that I have the firewall configured properly for VOIP QOS. What exactly have you done here? You do know that you are apparently using IAX2 and not SIP. Those are not the same protocols. In fact, if you configured the shorewall system for standard VoIP, that's your problem. IAX2 operates on different ports that SIP. Whereas SIP operates on a control port and then create media ports, IAX2 only uses one. As far as download speed, what have you told shorewall your download speed is? I'm not familiar with it, but just guessing that it's probably like most others. If this is the case, somewhere there is a setting to tell it what your download and upload speed is. 500kpbs up doesn't seem like enough bandwidth to support 10Mpbs down, either. ___ --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
[asterisk-users] Trying to register an G.729 codec boght from Digium and the register command does aboslutely nothing
Hello: I got into a trap. As far as I know I do not need to pay any royalties to use G.729b in Romania, so I should have used other drivers. The installation procedure looked difficult so I decided to get one from Digium - it's not that expensive, my time is much more expensive. Made the payment, got they key, downloaded and copied everything as in http://kb.digium.com/entry/30/5/; but when I called register I got no result. Actually I do get the prompt asking me to use -l to see the licence, nothing after that. It gives no error message, nothing at all! My first ethernet device is eth0 so it's not that; I'm able to browse https sites so the ports are open. I *disabled* the firewall and tried again, no success, so it's not firewall related. What to do next? 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] Buddy list order
If you are using the add on console then yes you can control it, but if you are
just using the phone then it will be in alphabetical order and not in the order
that you want. (I had this issue a month ago and as of then there was no fix
for this).
- Original Message -
From: Bryan M. Johns
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Wednesday, February 07, 2007 4:31 AM
Subject: Re: [asterisk-users] Buddy list order
Assuming you are using a central provisioning server, check your
{MAC}-directory.xml file. It contains the ordering that you are looking for.
I hope this helps.
Bryan M. Johns
Partner
Shelton | Johns Technology Group
office: 678:248:2637 x:1500
direct: 678:229:1809
mobile: 404.259.9216
iaxtel: 700:248:2637 x:1500
http://www.sheltonjohns.com
On Feb 6, 2007, at 9:35 PM, Bill Gibbs wrote:
I could have sworn I saw a post about this recently but I can’t find it so
apologies if this is a dupe, but is there anyway to control the order in the
Polycom Buddies list?
Bill
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RE: [asterisk-users] Using Local Channels with Originate
(Sorry for top-posting)
I'm making good progress. However, so as not to clutter the list I will
post my solution on the wiki in the next few days. I'll send out the
link as soon as I've got something substantial for you to review.
-MC
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian K.
Alexander,Jr. (Vision Point Systems)
Sent: Tuesday, February 06, 2007 6:10 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Using Local Channels with Originate
Ack... That should be I am using analog for the proof of concept but
plan to use PRI for the actual system...
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian K.
Alexander,Jr. (Vision Point Systems)
Sent: Tuesday, February 06, 2007 8:44 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Using Local Channels with Originate
Right now I am using analog but the plan is to use PRI for the proof of
concept but the actual system would use PRI. I know that the analog
support is supposed to be somewhat unreliable but I have yet to get it
to detect even a busy - not even once. I can only assume that I missed
some setting somewhere but I can't find it.
I am curious to learn more about your solution. If you post more
information I might be able to help you with your RD. In any event
thanks for posting up and in advance for keeping us posted on your
progress.
-Brian
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Collins
Sent: Monday, February 05, 2007 4:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Using Local Channels with Originate
I haven't quite gotten this working yet but I am going to update the
thread with what I have learned. Maybe this will help the next guy who
tries to figure this out...
The trick to using the DIALSTATUS seems to be to put it in the handler
for the h (hang-up extension).
[outdialer]
exten = 100, 1, Dial(${numberToDial})
exten = h, 1, Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,NoOp(Answered)
exten = s-BUSY,1,NoOp(Busy)
exten = s-NOANSWER,1,NoOp(Not answered)
exten = s-CANCEL,1,NoOp(Cancelled)
exten = s-CONGESTION,1,NoOp(Fast busy)
exten = s-CHANUNAVAIL,1,NoOp(Channel unavailable)
[dialerplan]
exten = s,1,Background(demo-congrats)
exten = s,n,WaitExten
so on ...
Here are the manager commands I am using:
Action: login
Username: test
Secret: nottelling
Action: originate
Channel: Local/[EMAIL PROTECTED]/n
Context: dialerplan
Extension: s
Priority: 1
Variable: numberToDial=ZAP/4/1234567890
Action: logoff
I am always getting ANSWERED for ${DIALSTAUS} so something is not quite
right. Hopefully I am getting closer.
Brian,
What kind of Zap hardware/telco lines are you using? I am using PRI and
I am able to get a dial status in the hangup extension. The problem I
run into is that I get NO ANSWER as the hangup cause even for invalid
phone numbers... I also get cluttered CDR's. In the meantime I'm
working on a solution that I hope will give the best of both worlds.
I'm relying on the API events instead of local channels. I'll post more
information when I've made more progress. However, I've made 2500 test
calls and I haven't lost a single 'OriginateSuccess' or
'OriginateFailure' event. (I'm keying on these, specifically the
'OriginateFailure' event because it has a 'Reason' value that gets
populated: 0=Invalid, 3=No Ans, 5=Busy.)
Hope to have more info posted this week.
-MC
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[asterisk-users] AMI Originate and release channels
Hi I set up call back functionally thru AMI (local channel). The two calls are bridged and the call is established. But when I hang up the local channel (the first extension that rang), the other leg of the call *is not released* Time events: 0) Socket communication(AMI) 1)extensionA ringing 2)extensionA picked up 3)extensionB ringing 4)extensionB picked up 5)Call is bridged 7)extensionA hangs up and released 8)extensionB still connected indefinitely I would like that extensionB was released when extensionA hangs up (When I first hang up extensionB the extensionA is released appropriately) Please, could you help me out with this issue. Thanks Paulo ___ --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] Trying to register an G.729 codec boght from Digium and the register command does aboslutely nothing
Cosmin Prund wrote: Hello: I got into a trap. As far as I know I do not need to pay any royalties to use G.729b in Romania, so I should have used other drivers. The installation procedure looked difficult so I decided to get one from Digium - it's not that expensive, my time is much more expensive. Made the payment, got they key, downloaded and copied everything as in http://kb.digium.com/entry/30/5/; but when I called register I got no result. Actually I do get the prompt asking me to use -l to see the licence, nothing after that. It gives no error message, nothing at all! My first ethernet device is eth0 so it's not that; I'm able to browse https sites so the ports are open. I *disabled* the firewall and tried again, no success, so it's not firewall related. What to do next? 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 Contact [EMAIL PROTECTED] -- One day at a time, one second if that's what it takes ___ --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] New Issue
On Wed, Feb 07, 2007 at 04:46:46PM +0200, Cosmin Prund wrote: the ./configure thing requires the sources of zaptel, Actually, it requires zaptel.h in the pointed place, or in the default place (as installed by the install target). Note that zaptel = 1.2 installs zaptel.h to /usr/include/linux, whereas zaptel 1.4 installs it to /usr/include/zaptel . -- 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
[asterisk-users] Can't get asterisk to compile chan_zap (was New Issue)
First, I didn't realize I hijacked another thread! Please accept my apologies. Now the problem: Asterisk isn't compiling chan_zap. chan_zap also doesn't appear in the list of channels when you make menuconfig I have read all the replies and specifically Cosmin's and Tzafrir's emails. zaptel.h is located in /usr/include/zaptel I also tried ./configure --with-zap=/usr/src/zaptel-1.4.0 which is the zaptel source and it didn't work either. I don't really know what else to try. I greatly appreciate any help TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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] Trying to register an G.729 codec boght from Digium and the register command does aboslutely nothing
So simple... I'm doing that right now, I've sent them an email. I didn't find that email address on Digium's support page... Thanks. Bruce Ferrell wrote: Cosmin Prund wrote: Hello: I got into a trap. As far as I know I do not need to pay any royalties to use G.729b in Romania, so I should have used other drivers. The installation procedure looked difficult so I decided to get one from Digium - it's not that expensive, my time is much more expensive. Made the payment, got they key, downloaded and copied everything as in http://kb.digium.com/entry/30/5/; but when I called register I got no result. Actually I do get the prompt asking me to use -l to see the licence, nothing after that. It gives no error message, nothing at all! My first ethernet device is eth0 so it's not that; I'm able to browse https sites so the ports are open. I *disabled* the firewall and tried again, no success, so it's not firewall related. What to do next? 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 Contact [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] Can't get asterisk to compile chan_zap (was NewIssue)
I had a typo in my last email. I meant --with-zaptel where I wrote --with-zap. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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] how to install Zaptel on Fedora linux 5
Hi, have a look at: http://www.aussievoip.com/wiki/index.php?page=freePBX-Centos This is based on Centos, but there is not a great difference between this and Fedora. It runs through all the requirements and installation for Zaptel and Asterisk in addition to the FreePBX web based config tool for Asterisk, which you can simply leave off if you do not want it. Hope this is of use, Robert Jenkins. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of A S Sent: 05 February 2007 21:10 To: asterisk-users@lists.digium.com Subject: [asterisk-users] how to install Zaptel on Fedora linux 5 Hi All, i have been trying in vain to fix the missing files with Zaptel installation for fedora 5 installation with asterisk 1.4.0. it will be great if someone can point me a to a good website with steps in it or give me some pointers on how to go about it. TIA, viopuser ___ --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] Can't get asterisk to compile chan_zap (was NewIssue)
I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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] Billing pulses
All digital lines (BRI or PRI) provides answer and release supervision. The drivers will send to * this information, and this information will be registered into the CDR automatically. You only need setup your billing system. As said before you do not need to intercept the billing pulse. Jorge Mendoza Stefano Corsi wrote: At 16.22 07/02/2007, you wrote: Funny that a digital line have a analogue pulse. Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system. Yes, it's strange. But I find no mention on answer supervision in the NT1Plus manual (NT1Plus is the hardware device the Telco installs when you ask for an ISDN line). Where should I ask for answer supervision? The Telco? That sounds very difficult in Italy... they have no technical call centers. Almost only sales. But if the line should provide those analog billing pulses... do you think could be possible to intercept them? Rgds Stefano ___ --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] Trying to register an G.729 codec boght from Digium and the register command does aboslutely nothing
On 7 Feb 2007, at 15:54, Cosmin Prund wrote: Hello: I got into a trap. As far as I know I do not need to pay any royalties to use G.729b in Romania, so I should have used other drivers. The installation procedure looked difficult so I decided to get one from Digium - it's not that expensive, my time is much more expensive. Made the payment, got they key, downloaded and copied everything as in http://kb.digium.com/entry/30/5/; but when I called register I got no result. Actually I do get the prompt asking me to use -l to see the licence, nothing after that. It gives no error message, nothing at all! My first ethernet device is eth0 so it's not that; I'm able to browse https sites so the ports are open. I *disabled* the firewall and tried again, no success, so it's not firewall related. What to do next? It may be firewall related, when I did it (a while back) registration tool used some 'non-standard' port I had to open for it. Still Digium support should help you there... Tim. 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 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
RE: [asterisk-users] Disconnection supervision: what about PBX
From:"Trevor G. Hammonds" [EMAIL PROTECTED] From: Yuan LIU Sent: Tuesday, February 06, 2007 8:11 PM After reading through several recent threads, I started to wonder why the Cisco document (and other VoIP documents) appears to present this issue as VoIP gateway specific.Don't (plain old) PBX' face the same issue if they use analogue interfaces?If there are analogue PBX' at all, how dothey solve the problem?Yuan,Well engineered analogue PBXs typically do not use standard loop startsubscriber lines.When digital trunks are not an option, they use analoguePBX and/or DID trunks.At the very least, ground start circuits arepreferred to avoid "glare".The best call quality for analogue is achievedby using four-wire EM trunks that provide answer and disconnectsupervision.There are two-wire trunks (which are probably more common), aswell as different signalling methods.These trunks require specialinterface hardware, and I am unaware of any that work with Asterisk.As thecards are typically very expensive, it is usually better to go with digitalif you require that functionality.It would be nice to see a BRI interfacefor Asterisk that works in North America, as BRI circuits are oftencomparable in price to analogue lines. Thanks for the enlightment, Trevor. I always thought that standard ISDN cards (presumably BRI) work with Asterisk if they work under Linux? Sincerely, Trevor Hammonds ___ --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] H323 to SIP - One way voice
Hello all, I want to use asterisk as protocol converter, H323 to SIP. I am using Asterisk 1.2.14 with chan_h323 and the free version of g729. When calling from SIP to H323 everything is fine. But when calling from H323 to SIP, the phone using SIP doesn't hear the other party. The phones and Asterisk are in the same subnet and the firewall of the Asterisk box is off. Please advice. Thank you, Andrei U ___ --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] s-${DIALSTATUS} extensions
From:"Eric \"ManxPower\" Wieling" [EMAIL PROTECTED]Yuan LIU wrote:In examples, s-${DIALSTATUS} is used to handle unsuccessful dial attempts in the s extension.Goto() is used in examples.Is the prefix "s-" mandatory? Is it related to the original extension "s"? (Apparently Goto(${DIALSTATUS}) won't work for me.)Goto(${DIALSTATUS}) won't work because with only one parameter Goto will think it is a priority.Try Goto(${DIALSTATUS},1)
Sorry for postingwrong code - was Goto(${DIALSTATUS},1). Probablythevery extensiondidn't get ${DIALSTATUS}, as others indicated that it only gets set when channel hangs up, in which case only extension h can be executed. On the other hand, the Asterisk "book" gave the example in extension s - haven't tested extensively either way, as this is not yet critical.
Yuan Liu
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Re: [asterisk-users] Something wrong with the list?
same for me, however today I started receiving the same amount as usual On 2/6/07, C F [EMAIL PROTECTED] wrote: Since Monday I didn't see much traffic. ___ --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] Something wrong with the list?
C F wrote: Since Monday I didn't see much traffic. gmail is having some sort of problem. I haven't gotten hardly any messages from any of the digium lists in my gmail account. ___ --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] Billing pulses
From:Jorge Mendoza [EMAIL PROTECTED]Funny that a digital line have a analogue pulse.Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system.Jorge MendozaStefano Corsi wrote:Hello,I've discovered that in Italy ISDN lines can be programmed to generate a "billing pulse" every n seconds (it dipends from the pricebook). The pulse has these figures: Whatever reason, if telco provides them, there's a good chance thatsome ISDN interface cards can use them. (Just googled to confirm that somenon-Digium cards can be used in Asterisk.) This doesn't mean that Asterisk can use them. So you may need significant programming to get going. Ifthey aretruly analogue pulses, it could be cheaper to produce a little dedicated circuit to feed an AGI or something. Yuan Liu frequency 12 kHz ?1%level .. 200 mVrms on 200distortion... 5%pulse duration .125 ?25 mspause duration 180 msperiod ... 300 msDoes someone know if these values can be used somehow to get an accurate billing using asterisk with these lines? Could be a matter of configuration or programming?ThanksStefano ___ --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] New Issue
In most cases, if I follow these steps, I get a working asterisk with zaptel: in asterisk, 1.4 and trunk: make distclean rm /usr/lib/asterisk/modules/* then, get the zaptel source that corresponds to your version of asterisk. configure, make, make install it as root. If you try to use a 1.4 zaptel with 1.2, prepare to be disappointed Then in 1.2 asterisk, do a make, or in 1.4/trunk, do a configure ( with --enable-dev-mode if you do this often), then make menuselect, make, make install (as root). Check the zaptel module stuff in menuselect. If you still don't have zaptel, consult your config.log, and look for anything about zaptel. It might be informative. It should be finding zaptel stuff. murf On Wed, 2007-02-07 at 18:43 +0200, Tzafrir Cohen wrote: On Wed, Feb 07, 2007 at 04:46:46PM +0200, Cosmin Prund wrote: the ./configure thing requires the sources of zaptel, Actually, it requires zaptel.h in the pointed place, or in the default place (as installed by the install target). Note that zaptel = 1.2 installs zaptel.h to /usr/include/linux, whereas zaptel 1.4 installs it to /usr/include/zaptel . -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ --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] Diagnosing poor call quality
Greetings list, We have an issue with call quality at 2 sites where the users (4 Elmeg IP290s at one site, 2 SPA942s at the other) do not have an asterisk box on-site. Each site has an 8mb down/448k up ADSL connection and the phones connect via SIP to an asterisk box in a datacentre using g729. The asterisk box in the datacentre connects to our other asterisk boxes providing pstn connectivity via IAX2. Latency between these boxes is between 1 and 2ms. The ADSL connections to the client sites are all consistently delivering latencies of sub-25ms to the datacentre and there is traffic shaping on that connection to give priority to any traffic from the phones' IPs. Comments from the users at these sites are as follows: call sounded like a dalek and I couldn't make out anything the caller was saying the phone on my desk is breaking up so badly it's virtually unusable calls sound like they're breaking up with metallic background noises We have quite a few customers with asterisk boxes on-site (with phones connected to them via the LAN) using ADSL connections from the same supplier, and are not having these issues with them. canreinvite=no and nat=yes are set on all these devices, since they are behind NAT. Each device re-registers with asterisk every 5 minutes to prevent any possible NAT state timeouts. Any pointers/places to look for potential problems would be much appreciated. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited 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
RE: [asterisk-users] Having Trouble With Wait Command in CallbackContext
From:"Robert DeVries" [EMAIL PROTECTED] I am trying to get called back with a DISA dial tone when I call a trigger number. I got it to work almost the way I want, this is the callback context:[callback]exten= 501,1,Congestion() exten= 501,2,Hangup() exten =h,1,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing)exten =h,2,Hangup() With the above, the call comes into the trigger number, then the call file is copied and executed, I get the DISA dial tone, and can dial just fine. However, the problem is that the callback is a bit too fast, and sometimes calls back before I can hang up, even if I hang up fast. I want to program in a pause. However, when I do the following:exten= 501,1,Congestion() exten= 501,2,Hangup() exten =h,1,wait (10) exten =h,2,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) exten =h,3,Hangup() the callback never occurs, the execution never gets beyond the wait command.So, two questions - why does it not execute once I insert the wait command, and how do I get a wait before the call file is run. People who know will answer the first question. But the second question has a ready answer: if you are using System(), why not insert a sleep right there? exten =h,1,System(sleep 10; cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) Hope this helps. Yuan Liu ___ --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] Softphone on Linux
please send me more info thanks! Tim Panton [EMAIL PROTECTED] wrote: On 5 Feb 2007, at 21:46, chester c young wrote: Need to deploy between 50 to 300 lightweight Linux - only browser and softphone. You might want to consider our lightweight java softphone (Corraleta SDK) - it can be embedded in a web page - zero install/config in the client. The UI is in HTML and javascript, so you can get it _exactly_ the way you want it. Any recomendations? Clearly I'm biased :-) Tim Panton www.mexuar.com 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 - Everyone is raving about the all-new Yahoo! Mail beta.___ --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] OpenSuSE Firewall2 - Traffic Shaping
Has anyone got any hints of how to setup the OpenSuSE Firewall2 with VOIP friendly traffic shaping? The only bit I see is in the config file regarding how to setup a simple HTB. I come from Shorewall, and am finding this firewall to be different. Any help is appreciated. Actually, either Shorewall and SuSEfirewall2 use linux iptables package filter. So, you can disable SuSEfirewall2 and install Shorewall if you feel more confy with it. I havent found a specific package for suse, but only generic rpms. -- Saludos, miguel Los agujeros negros son lugares donde dios dividió por cero. Black holes are places where god divided by zero. ___ --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] Can't get asterisk to compile chan_zap (was NewIssue)
(I apologize if this is a dupe, but I never saw my first copy) I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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] Can't get asterisk to compile chan_zap (was NewIssue)
David Ruggles wrote: I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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 I didnt follow your original thread... chan_zap does not appear in menuselect? Does it exist in channels directory? ___ --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] semi-private call
Hi Do you know if the SIP protocol is compatible with semi-private calls. I can contruct a private call by putting the SIP Privacy header to id and then sending the call to my SIP-Pri box and it works This tell my Pri provider that the call is private. How can I tell my Pri provider that the call is semi-private ? Patrick ___ --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] Can't get asterisk to compile chan_zap (was NewIssue)
David Ruggles wrote: I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [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 check too that menuselect-tree has an entry for chan_zap (it's in source root) ___ --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] Disconnection supervision: what about PBX
(Previous reply got garbled in Hotmail) From:"Trevor G. Hammonds" [EMAIL PROTECTED]Date:Wed, 7 Feb 2007 04:49:08 -0800 From: Yuan LIU Sent: Tuesday, February 06, 2007 8:11 PM After reading through several recent threads, I started to wonder why the Cisco document (and other VoIP documents) appears to present this issue as VoIP gateway specific.Don't (plain old) PBX' face the same issue if they use analogue interfaces?If there are analogue PBX' at all, how do they solve the problem?Yuan,Well engineered analogue PBXs typically do not use standard loop startsubscriber lines.When digital trunks are not an option, they use analoguePBX and/or DID trunks.At the very least, ground start circuits arepreferred to avoid "glare".The best call quality for analogue is achievedby using four-wire EM trunks that provide answer and disconnectsupervision.There are two-wire trunks (which are probably more common), aswell as different signalling methods.These trunks require specialinterface hardware, and I am unaware of any that work with Asterisk.As thecards are typically very expensive, it is usually better to go with digitalif you require that functionality.It would be nice to see a BRI interfacefor Asterisk that works in North America, as BRI circuits are oftencomparable in price to analogue lines. Thanks for enlightenment, Trevor. I always thought Eicon and other standard BRI cards work with Asterisk? Yuan Liu Sincerely, Trevor Hammonds ___ --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] Can't get asterisk to compile chan_zap (wasNewIssue)
From:"David Ruggles" [EMAIL PROTECTED]Date:Wed, 7 Feb 2007 12:15:37 -0500I captured the output of ./configure and found the following lines:lines snippedchecking zaptel/tonezone.h usability... yeschecking zaptel/tonezone.h presence... yeschecking for zaptel/tonezone.h... yeslines snippedchecking for ZT_TONE_DTMF_BASE in zaptel.h... Yeslines snippedSo it seems to be finding the /usr/include/zaptel directory and files fine.Is there anything else I can do that might offer information that could helptrack this problem down? At least one version of Asterisk (1.4?) requires correctkernel driver configuration before compilation. Have you done ztconfig and stuff? Yuan Liu Thanks,David RugglesCCNA MCSE (NT) CNA A+Network Engineer Safe Data, Inc.(910) 285-7200 [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] CPU motherboard for 100+ simultaneouse calls on Digium Quad E1 TE411p
hi! anyone please recommend/guide me of purchasing a resonably high performance server system regarding processor(s) motherboard (+ other compulsary peripherals i.e. VGA, Soundcard). Mentioning up-to-date vendor+model will be more helping I've to use Digium TE411p Quad E1 card signalling on the E1 is SS7 no. of simultaneouse calls is 100+ regards ___ --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: Help - Poor Voice Quality
Tim, What sort of 'poor' quality are we talking about - when folks complain what words do they use? On the other end, folks complain that the voice drops out. Words are lost. It's very frustrating to communicate. Which codec(s) are you using? ULAW How many channels do you want to use at once ? 1 is fine. This is basic home use. What is the round-trip time between you and the teliax server ? The ping responses are on the order of 15mS. I ran mtr, teliax is 10 hops away, and I don't see any packet loss. I was just on the phone with my house, and the call sounded just fine at this time (problems come and go). This is the dump of iax2 show netstats while the call was up. LOCAL - REMOTE ChannelRTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/teliax-2 37 -10-1 -1 0 -1 50 40 0 0 00 0 Do you have the jitterbuffer on or off ? I don't believe so. I didn't turn jitter on. I believe jitter is off by default. Thanks, Jim Tim Panton [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] On 7 Feb 2007, at 03:59, Jim Duda wrote: Thanks for the reply Lacy. Yes, I know that I am using IAX2 and not SIP for my connection to teliax. IAX2 is the preferred protocol for connection to teliax. I have the firewall configured to prioritorize port 4569 for IAX2. I have the shorewall tcdevices file setup with 3 mbit download and 500 Kbit upload speeds. We need a few clues :-) What sort of 'poor' quality are we talking about - when folks complain what words do they use? Which codec(s) are you using? How many channels do you want to use at once ? What is the round-trip time between you and the teliax server ? Do you have the jitterbuffer on or off ? (if you only have 6ms of jitter, I'd switch it off) 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 ___ --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: Help - Poor Voice Quality
Yes, I had seen something in various posts about using SIP instead of IAX2. I have been switching back and forth between IAX2 and SIP, however, I haven't seen any noticeable difference. I will try a switch back to SIP again and see how that goes. Jim James Fromm [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Jim, I too am a Teliax user. Talk to their technical support. IAX2 is NOT preferred. They'll tell you to use SIP. Jim Duda wrote: Thanks for the reply Lacy. Yes, I know that I am using IAX2 and not SIP for my connection to teliax. IAX2 is the preferred protocol for connection to teliax. I have the firewall configured to prioritorize port 4569 for IAX2. I have the shorewall tcdevices file setup with 3 mbit download and 500 Kbit upload speeds. Jim Lacy Moore wrote: Jim Duda wrote: I've been on the shorewall firewall and confirmed that I have the firewall configured properly for VOIP QOS. What exactly have you done here? You do know that you are apparently using IAX2 and not SIP. Those are not the same protocols. In fact, if you configured the shorewall system for standard VoIP, that's your problem. IAX2 operates on different ports that SIP. Whereas SIP operates on a control port and then create media ports, IAX2 only uses one. As far as download speed, what have you told shorewall your download speed is? I'm not familiar with it, but just guessing that it's probably like most others. If this is the case, somewhere there is a setting to tell it what your download and upload speed is. 500kpbs up doesn't seem like enough bandwidth to support 10Mpbs down, either. ___ --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] Something wrong with the list?
I saw the same thing, but got a huge flood of messages today. A Gmail issue perhaps? Alex On 2/6/07, C F [EMAIL PROTECTED] wrote: Since Monday I didn't see much traffic. ___ --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 -- Alex Robar [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] Zaptel bug
Hi all, Is anyone aware of any progress on this bug? http://bugs.digium.com/view.php?id=8763 Not only is the channel randomly disappearing during idle periods, it vanishes during a call as well. No indications in dmesg, syslog, asterisk or anything. Only cure is to rmmod and modprobe again. I'm currently on 1.4.0. Any ideas would be greatly appreciated. Cheers, Kyle -- Kyle Gordon [EMAIL PROTECTED] http://lodge.glasgownet.com pgpg0dWiykWpB.pgp Description: PGP signature ___ --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] List problem handling HTML E-mails?
My multiple postings to this list this morning got garbled in http://lists.digium.com/pipermail/asterisk-users/, and don't come back from list. (e.g., http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I thought it was Hotmail, so I saved one outgoing mail and checked that it's correct. Anyone else experiencing same? Yuan Liu ___ --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: Linksys auto provision
Found my answer for those who would like to know: Profile Rule: [--key $A]http://your.addre.ss/$B/$MAC.cfg GPP A: urtopsecretultrasecureaesencryptionkey GPP B: OddBallDirectory123098 Hope that helps someone! Curt -Original Message- From: Curt Shaffer [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 07, 2007 11:35 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Linksys auto provision I have a question about encrypted configs for the Linksys device auto configuration. I am able to do it with xml no problem. However when I generate the text file with the SPC tool then encrypt it with the tool the settings do not take affect. The ATA grabs the correct file but nothing I change is modified when it gets the new config. My guess is that the ATA needs to have the passphrase for the encryption somewhere but none of the fields appear to be labeled passphrase or something intuitive to know where to put it. Any help is appreciated! Thanks! Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] List problem handling HTML E-mails?
My multiple postings to this list this morning got garbled in http://lists.digium.com/pipermail/asterisk-users/, and don't come back from list. (e.g., http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I thought it was Hotmail, so I saved one outgoing mail and checked that it's correct. Anyone else experiencing same? Yuan Liu ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] s-${DIALSTATUS} extensions
Yuan LIU wrote:
From: /Eric \ManxPower\ Wieling [EMAIL PROTECTED]/
Yuan LIU wrote:
In examples, s-${DIALSTATUS} is used to handle unsuccessful dial
attempts in the s extension. Goto() is used in examples. Is the
prefix s- mandatory? Is it related to the original extension s?
(Apparently Goto(${DIALSTATUS}) won't work for me.)
Goto(${DIALSTATUS}) won't work because with only one parameter Goto
will think it is a priority. Try Goto(${DIALSTATUS},1)
Sorry for posting wrong code - was Goto(${DIALSTATUS},1).
Probably the very extension didn't get ${DIALSTATUS}, as others
indicated that it only gets set when channel hangs up, in which case
only extension h can be executed. On the other hand, the Asterisk
book gave the example in extension s - haven't tested extensively
either way, as this is not yet critical.
DIALSTATUS and HANGUPCAUSE are most useful for situations where the call
was NOT answered and you want to know why.
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Re: [asterisk-users] Something wrong with the list?
From: Lacy Moore [EMAIL PROTECTED] Date: Wed, 07 Feb 2007 12:10:01 -0600 C F wrote: Since Monday I didn't see much traffic. gmail is having some sort of problem. I haven't gotten hardly any messages from any of the digium lists in my gmail account. It's the list, not gmail. Check dates in mails received today. I even got many double mails. (I tried password reminder, unsubscribe, resubscribe, on Monday. Nothing worked then.) Yuan Liu ___ --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] Billing pulses
Hi Billing Pulses only apply to analogue lines. You need special hardware in the PBX interface to detect them and pass them on to the Billing software. To my knowlege there is no Asterisk compatible hardware that does this. George - Original Message - From: Stefano Corsi [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, February 07, 2007 4:04 PM Subject: [asterisk-users] Billing pulses Hello, I've discovered that in Italy ISDN lines can be programmed to generate a billing pulse every n seconds (it dipends from the pricebook). The pulse has these figures: frequency 12 kHz ± 1% level .. 200 mVrms on 200 distortion... 5% pulse duration .125 ± 25 ms pause duration 180 ms period ... 300 ms Does someone know if these values can be used somehow to get an accurate billing using asterisk with these lines? Could be a matter of configuration or programming? Thanks Stefano ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.29/673 - Release Date: 2/6/2007 5:52 PM ___ --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] Softphone +Realtime
Here's an interesting issue we're facing... We would like users to be able to use softphones from home/work and to use their same extensions they do at work. The first step of getting the phones to log in as their same extensions as work is easy and works. However, on the database side, once the client closes, the sip table is cleared of the ip to the phone. This means that no calls are forwarded to their office line anymore, and instead have to just go to voicemail. To fix this, the best I can think of is to replace those values nightly and update the timestamp so asterisk knows to update its values. Has anyone tried anything like this? I would like the phones to regrab their spot once the softphone is logged out. We have a Asterisk box (gentoo linux) which is running realtime (mysql 5). Our phones are Polycom SoundPoint 501s and the softphone is xlite (windows). Rob ___ --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] Diagnosing poor call quality
Chris Bagnall wrote: Greetings list, We have an issue with call quality at 2 sites where the users (4 Elmeg IP290s at one site, 2 SPA942s at the other) do not have an asterisk box on-site. Each site has an 8mb down/448k up ADSL connection and the phones connect via SIP to an asterisk box in a datacentre using g729. The asterisk box in the datacentre connects to our other asterisk boxes providing pstn connectivity via IAX2. Latency between these boxes is between 1 and 2ms. The ADSL connections to the client sites are all consistently delivering latencies of sub-25ms to the datacentre and there is traffic shaping on that connection to give priority to any traffic from the phones' IPs. Comments from the users at these sites are as follows: call sounded like a dalek and I couldn't make out anything the caller was saying the phone on my desk is breaking up so badly it's virtually unusable calls sound like they're breaking up with metallic background noises We have quite a few customers with asterisk boxes on-site (with phones connected to them via the LAN) using ADSL connections from the same supplier, and are not having these issues with them. canreinvite=no and nat=yes are set on all these devices, since they are behind NAT. Each device re-registers with asterisk every 5 minutes to prevent any possible NAT state timeouts. Any pointers/places to look for potential problems would be much appreciated. This should be a FAQ. Set the RTP packet size on the SPAs to .2 instead of .3 ___ --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] After upgrade to 1.4 transfers don't work properly
I have discovered an issue on my system after upgrading from 1.2.13 to 1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone. I have confirmed this on multiple phones. When the called person answers and tries to transfer the call to another extension, the call successfully transfers, however the person answering the transfer cannot hear the person that called in, the caller. My dial command simply is exten=4000,1,Dial(SIP/4000,40,t) This DID work before when we were on 1.2.13. The CLI displays the following message: handle_response: Notify answer on an owned channel? I searched the web and found similar issues but not the same. http://bugs.digium.com/view.php?id=8696 This one has the error, however I don't get a segment fault and supposedly this was fixed in revision 50032. How do I get this revision? I'm guessing it's in a non-tested svn release which I don't think I want to install in a production system. Anyone else have this issue? Any ideas on how to fix this or get around it? _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com http://www.novo1.com/ Novo 1 is a service mark of Novo 1, 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] Can't get asterisk to compile chan_zap (wasNewIssue)
(Hotmail garbled reply again) From:"David Ruggles" [EMAIL PROTECTED]Date:Wed, 7 Feb 2007 12:15:37 -0500I captured the output of ./configure and found the following lines:lines snippedchecking zaptel/tonezone.h usability... yeschecking zaptel/tonezone.h presence... yeschecking for zaptel/tonezone.h... yeslines snippedchecking for ZT_TONE_DTMF_BASE in zaptel.h... Yeslines snippedSo it seems to be finding the /usr/include/zaptel directory and files fine.Is there anything else I can do that might offer information that could helptrack this problem down? AtleastoneversionofAsterisk(1.4?)requirescorrectnbsp;kerneldriverconfigurationbeforecompilation. Haveyoudoneztconfigandstuff? Yuan Liu Thanks,David RugglesCCNA MCSE (NT) CNA A+Network Engineer Safe Data, Inc.(910) 285-7200 [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] Billing pulses
On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote: From: Jorge Mendoza [EMAIL PROTECTED] Funny that a digital line have a analogue pulse. Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system. Jorge Mendoza Stefano Corsi wrote: Hello, I've discovered that in Italy ISDN lines can be programmed to generate a billing pulse every n seconds (it dipends from the pricebook). The pulse has these figures: Whatever reason, if telco provides them, there's a good chance that some ISDN interface cards can use them. (Just googled to confirm that some non-Digium cards can be used in Asterisk.) This doesn't mean that Asterisk can use them. So you may need significant programming to get going. If they are truly analogue pulses, it could be cheaper to produce a little dedicated circuit to feed an AGI or something. Yuan Liu frequency 12 kHz ?1% level .. 200 mVrms on 200 distortion... 5% pulse duration .125 ? 25 ms pause duration 180 ms period ... 300 ms Does someone know if these values can be used somehow to get an accurate billing using asterisk with these lines? Could be a matter of configuration or programming? Thanks Stefano ___ --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 How would you be able to determine which call was being billed for if the pulse is sent down the wire on an ISDN circuit with multiple channels in use? db ___ --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] Softphone on Linux
On 10:43, Wed 07 Feb 07, chester c young wrote: please send me more info thanks! Tim Panton [EMAIL PROTECTED] wrote: On 5 Feb 2007, at 21:46, chester c young wrote: Need to deploy between 50 to 300 lightweight Linux - only browser and softphone. You might want to consider our lightweight java softphone (Corraleta SDK) - it can be embedded in a web page - zero install/config in the client. The UI is in HTML and javascript, so you can get it _exactly_ the way you want it. Is this a commercial app? I cant find a download link or something about prices. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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] Can't get asterisk to compile chan_zap (wasNewIssue)
I've been trying to snip message to keep them from getting too large, maybe I over did it. :) chan_zap.c is in /usr/src/asterisk-1.4.0/channels But doesn't show up in the list of channels in make menuconfig Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Gonzalez Sent: Wednesday, February 07, 2007 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get asterisk to compile chan_zap (wasNewIssue) David Ruggles wrote: I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[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 I didnt follow your original thread... chan_zap does not appear in menuselect? Does it exist in channels directory? ___ --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] Diagnosing poor call quality
The advertised datarate (8mb/448k) are the speeds at which the circuit between the customer and the central office is clocked and has no relationship with *effective* throughput. At the central office are *shared* facilities than connects each DSL connection with the network, and over subscription to these shared facilities cause congestion. Also, there is no QoS on the Internet, and congestion anywhere between the end points will cause poor call quality. Disclaimer: The following information is several months old--I've since moved my customers away from Qwest DSL. Here in Denver we have Qwest DSL service from a central office where the effective throughput drops to dialup speeds during the day. Regular web/email users don't usually notice packet loss because dropped packets are recovered by the TCP protocol. For VoIP on UDP, however, the call quality suffers to the point of being unusable (clicking, popping, and dropouts). Furthermore, Qwest doesn't have Denver peering with the rest of the Internet. To leave the Qwest network, connections typically go to DAL, LAX, or SFO on congested circuits. So beware of VoIP over DSL. Your users need to be aware of the tradeoffs between the cost of DSL vs. T1 and the effect on call quality. Chris, if your customers are in the western US then please contact me about dedicated circuits. Chris Bagnall wrote: Greetings list, We have an issue with call quality at 2 sites where the users (4 Elmeg IP290s at one site, 2 SPA942s at the other) do not have an asterisk box on-site. Each site has an 8mb down/448k up ADSL connection and the phones connect via SIP to an asterisk box in a datacentre using g729. The asterisk box in the datacentre connects to our other asterisk boxes providing pstn connectivity via IAX2. Latency between these boxes is between 1 and 2ms. The ADSL connections to the client sites are all consistently delivering latencies of sub-25ms to the datacentre and there is traffic shaping on that connection to give priority to any traffic from the phones' IPs. Comments from the users at these sites are as follows: call sounded like a dalek and I couldn't make out anything the caller was saying the phone on my desk is breaking up so badly it's virtually unusable calls sound like they're breaking up with metallic background noises We have quite a few customers with asterisk boxes on-site (with phones connected to them via the LAN) using ADSL connections from the same supplier, and are not having these issues with them. canreinvite=no and nat=yes are set on all these devices, since they are behind NAT. Each device re-registers with asterisk every 5 minutes to prevent any possible NAT state timeouts. Any pointers/places to look for potential problems would be much appreciated. Regards, Chris ___ --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] Can't get asterisk to compile chan_zap (wasNewIssue)
Menuselect-tree does have a member entry for chan_zap. I has two depend subnodes and one use subnode. The depends are: zaptel and tonezone The use is: pri (I've installed libpri also) Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rodrigo Gonzalez Sent: Wednesday, February 07, 2007 2:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get asterisk to compile chan_zap (wasNewIssue) David Ruggles wrote: I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[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 check too that menuselect-tree has an entry for chan_zap (it's in source root) ___ --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] Can't get asterisk to compile chan_zap(wasNewIssue)
that I have! :) Have a single X100P in the system and ztcfg configures the board no problem. zttool confirms the board is there and shows RED when the phone line is removed and OK when the phone line is plugged in. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200[EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU Sent: Wednesday, February 07, 2007 2:02 PM To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] Can't get asterisk to compile chan_zap(wasNewIssue) From: David Ruggles [EMAIL PROTECTED] Date: Wed, 7 Feb 2007 12:15:37 -0500 I captured the output of ./configure and found the following lines: lines snipped checking zaptel/tonezone.h usability... yes checking zaptel/tonezone.h presence... yes checking for zaptel/tonezone.h... yes lines snipped checking for ZT_TONE_DTMF_BASE in zaptel.h... Yes lines snipped So it seems to be finding the /usr/include/zaptel directory and files fine. Is there anything else I can do that might offer information that could help track this problem down? At least one version of Asterisk (1.4?) requires correct kernel driver configuration before compilation. Have you done ztconfig and stuff? Yuan Liu Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [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] CPU motherboard for 100+ simultaneouse calls onDigium Quad E1 TE411p
Which codec do you plan to use? Henk From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of umar tarar Sent: woensdag 7 februari 2007 20:05 To: asterisk-users@lists.digium.com Subject: [asterisk-users] CPU motherboard for 100+ simultaneouse calls onDigium Quad E1 TE411p hi! anyone please recommend/guide me of purchasing a resonably high performance server system regarding processor(s) motherboard (+ other compulsary peripherals i.e. VGA, Soundcard). Mentioning up-to-date vendor+model will be more helping I've to use Digium TE411p Quad E1 card signalling on the E1 is SS7 no. of simultaneouse calls is 100+ regards ___ --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] List problem handling HTML E-mails?
On Wed, Feb 07, 2007 at 11:45:30AM -0800, Yuan Liu said: My multiple postings to this list this morning got garbled in http://lists.digium.com/pipermail/asterisk-users/, and don't come back from list. (e.g., http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I thought it was Hotmail, so I saved one outgoing mail and checked that it's correct. Anyone else experiencing same? It's bad netiquette to send HTML to mailing lists in general. It hoses up digests and archives, and some people don't have HTML capable clients. Some mail clients send both plain text and html, which isn't quite so bad since the receipient / archiving software can pick out the plain text version, but clients that send HTML ONLY should be avoided. Check to see if you can configure your mail service to use plain text and that should fix things. ___ --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] Mysterious tables starting with stats_
Is there any sort of friendly interface installed on that box? []'s MM -Original Message- From: José Pablo Fernández [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Cc: Sent: Tue, 6 Feb 2007 17:32:59 -0300 Delivered: Tue, 06 Feb 2007 16:44:11 Subject:[asterisk-users] Mysterious tables starting with stats_ I have a server which I haven't installed that I have to maintain. This server uses MySQL, it has an asterisk database and in there some mysterious tables: stats_action, stats_agent, stats_callid, stats_config, stats_estados, stats_qstats, stats_queue, stats_queuexagent. I say mysterious because I don't have a clue about who is generating them. I did a grep for their names in /var, /root and /etc with no luck. Does anybody have any ideas what might be generating them, and populating them? Thanks. -- José Pablo Fernández [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 E-mail classificado pelo Identificador de Spam Inteligente Terra. Para alterar a categoria classificada, visite http://mail.terra.com.br/protected_email/imail/imail.cgi?+_u=levelz_l=1,1170794651.149490.15840.alcuta.terra.com.br,3903,Des15,Des15 --Original Message Ends-- -- Melcon Moraes [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] Billing pulses
On Wed, 2007-02-07 at 21:00 +0100, George Camilleri wrote: Hi Billing Pulses only apply to analogue lines. You need special hardware in the PBX interface to detect them and pass them on to the Billing software. To my knowlege there is no Asterisk compatible hardware that does this. ISDN has AOC (advice of charge) and does not require special hardware. Iirc a while back there was some development of AOC support for Asterisk but I am not aware of the current status. Regards, Patrick ___ --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] Billing pulses
From: David Boyd [EMAIL PROTECTED] Date: Wed, 07 Feb 2007 15:24:04 -0500 On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote: From: Jorge Mendoza [EMAIL PROTECTED] Funny that a digital line have a analogue pulse. Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system. Jorge Mendoza Stefano Corsi wrote: Hello, I've discovered that in Italy ISDN lines can be programmed to generate a billing pulse every n seconds (it dipends from the pricebook). The pulse has these figures: Whatever reason, if telco provides them, there's a good chance that some ISDN interface cards can use them. (Just googled to confirm that some non-Digium cards can be used in Asterisk.) This doesn't mean that Asterisk can use them. So you may need significant programming to get going. If they are truly analogue pulses, it could be cheaper to produce a little dedicated circuit to feed an AGI or something. Yuan Liu ... How would you be able to determine which call was being billed for if the pulse is sent down the wire on an ISDN circuit with multiple channels in use? db Bill them both. We are talking about mere BRI's, right:-) Good catch, David. As others noted, billing pulse really applies to analogue lines only, and ISDN providers should always send status. Yuan Liu ___ --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] does any one knows of a Softphone that works under terminal services?
Hi all I'm looking for a softphone that works well under terminal services environment, we need to set up 24 to 32 phones for a call center, also, does any one knows if it will actually work fine under load? ___ --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] Softphone on Linux
Hi Michiel, Yes it's a commercial app; all the info you need is on the wiki including pricing and installation guide. http://www.voip-info.org/wiki/view/Mexuar Feel free to send me an email if you are the USA with any questions or Tim if you are in the UK (talk about round the world support - answering his email as he has probably logged off for the evening :) . Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michiel van Baak Sent: Wednesday, 7 February 2007 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Softphone on Linux On 10:43, Wed 07 Feb 07, chester c young wrote: please send me more info thanks! Tim Panton [EMAIL PROTECTED] wrote: On 5 Feb 2007, at 21:46, chester c young wrote: Need to deploy between 50 to 300 lightweight Linux - only browser and softphone. You might want to consider our lightweight java softphone (Corraleta SDK) - it can be embedded in a web page - zero install/config in the client. The UI is in HTML and javascript, so you can get it _exactly_ the way you want it. Is this a commercial app? I cant find a download link or something about prices. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called 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
[asterisk-users] Red alarms
Asterisk is getting red alarms on my T1, sometimes once or twice a day, but today it happened 5 times. Even once is too many. Every call in progress is dropped. Please help! What do I need to do? What can I try? I've googled and searched this list and can't find anything. Here's an example from the logs: Feb 7 13:37:54 WARNING[32451] chan_zap.c: Detected alarm on channel 6: Red Alarm Feb 7 13:37:54 WARNING[32238] chan_zap.c: Detected alarm on channel 1: Red Alarm Feb 7 13:37:54 WARNING[32546] chan_zap.c: Detected alarm on channel 2: Red Alarm Feb 7 13:37:54 WARNING[32477] chan_zap.c: Detected alarm on channel 3: Red Alarm Feb 7 13:37:54 WARNING[32537] chan_zap.c: Detected alarm on channel 4: Red Alarm Feb 7 13:37:54 WARNING[32557] chan_zap.c: Detected alarm on channel 5: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 7: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 9: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 11: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 12: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 13: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 14: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 15: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 16: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 17: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 18: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 19: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 20: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 21: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 22: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 23: Red Alarm Feb 7 13:37:54 WARNING[2524] chan_zap.c: Detected alarm on channel 24: Red Alarm Feb 7 13:37:54 WARNING[32566] chan_zap.c: Detected alarm on channel 8: Red Alarm Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 1 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 2 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 3 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 4 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 5 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 6 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 7 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 8 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 9 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 10 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 11 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 12 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 13 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 14 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 15 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 16 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 17 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 18 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 19 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 20 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 21 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 22 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 23 Feb 7 13:37:59 NOTICE[2524] chan_zap.c: Alarm cleared on channel 24 Here are other items that appear in the logs quite frequently that I think are related (and probably all related to timing): Feb 7 15:52:13 WARNING[8920] chan_zap.c: Ring/Off-hook in strange state 6 on channel 4 Feb 7 15:52:21 WARNING[2524] chan_zap.c: zt hook failed: Device or resource busy We also get ghost calls from time to time. More info: EM Wink, B8ZS, ESF Digium Wildcard TE405P The telco told me that timing must be provided by us, but when I tried that all hell broke loose. ___ --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] After upgrade to 1.4 transfers don't work properly
On Wed, 2007-02-07 at 14:12 -0600, Savoy, Kevin - Williston, ND wrote: I have discovered an issue on my system after upgrading from 1.2.13 to 1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone. I have confirmed this on multiple phones. When the called person answers and tries to transfer the call to another extension, the call successfully transfers, however the person answering the transfer cannot hear the person that called in, the caller. My dial command simply is I had exactly the same problem when upgrading to 1.4 and I solved by making sure canreinvite=no is in sip.conf for every phone. -- Telecomunicaciones Abiertas de Mexico S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Having Trouble With Wait Command in CallbackContext
I just tried what you suggested - it executes the sleep for 10 seconds, then skips down to the hangup, without copying the call file to begin the callback. However, I then broke the system command into two lines like this: exten =h,1,System(sleep 10) exten =h,2,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) and it worked perfectly. Problem solved. Thanks. On 2/7/07, Yuan LIU [EMAIL PROTECTED] wrote: From: *Robert DeVries [EMAIL PROTECTED] I am trying to get called back with a DISA dial tone when I call a trigger number. I got it to work almost the way I want, this is the callback context: [callback] exten= 501,1,Congestion() exten= 501,2,Hangup() exten =h,1,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) exten =h,2,Hangup() With the above, the call comes into the trigger number, then the call file is copied and executed, I get the DISA dial tone, and can dial just fine. However, the problem is that the callback is a bit too fast, and sometimes calls back before I can hang up, even if I hang up fast. I want to program in a pause. However, when I do the following: exten= 501,1,Congestion() exten= 501,2,Hangup() exten =h,1,wait (10) exten =h,2,System(cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) exten =h,3,Hangup() the callback never occurs, the execution never gets beyond the wait command. So, two questions - why does it not execute once I insert the wait command, and how do I get a wait before the call file is run. People who know will answer the first question. But the second question has a ready answer: if you are using System(), why not insert a sleep right there? exten =h,1,System(sleep 10; cp /etc/asterisk/callback.info /var/spool/asterisk/outgoing) Hope this helps. Yuan Liu ___ --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] Test to Speech
Someone has worked with any test to speech software with aceptable quality in spanish? Probably in english the text to speech quality will be better. Witch test to speech software gave you the best results in spanish? Hi Andres, Check www.loquendo.com out... They have a nice web front end for demoing their product's abilites. Good Luck, -- Ex Vito ___ --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] Diagnosing poor call quality
Eric said: This should be a FAQ. Set the RTP packet size on the SPAs to .2 instead of .3 Thanks for the suggestion. I've logged into the offending devices and set both to .2. I'll see how it goes for 48 hours or so. I've looked at the Elmeg ip290's and they are set to 20ms from factory, so I don't think that's the issue there. Michael said: The advertised datarate (8mb/448k) are the speeds at which the circuit between the customer and the central office is clocked and has no relationship with *effective* throughput. I have run a few speed tests from the sites in question (iperf to the machine in the datacentre) and I'm consistently getting around 380k upstream and 5.5mbit downstream, even during peak hours. Some distance away from the quoted speeds, but still plenty enough to support 4 SIP devices using g729 (which should be about 30kbit/sec per device including packet overheads). So beware of VoIP over DSL. Your users need to be aware of the tradeoffs between the cost of DSL vs. T1 and the effect on call quality. Alas, T1 for net traffic here in the UK is insanely expensive. DSL in its various forms is about the best we get, and SDSL with low contention ratios (1:1, 5:1, etc.) is only available in a few exchanges in major cities. Chris, if your customers are in the western US then please contact me about dedicated circuits. About 4500 miles away. :-) Thanks for the offer anyway though. Any further thoughts would be gratefully appreciated, especially for the site with the Elmegs. Regards, Chris -- C.M. Bagnall, Director, Minotaur I.T. Limited 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
Re: [asterisk-users] Sending sound to an open channel....
In a dialplan, after i set an autohangup (with AGI) , how could i send a sound (stream a sound ) into an open channel at X seconds before the autohangup time get to 0 for that channel? (Like public phones, that gives u a 'beep!!!' before ur time runs out, just like that...) Check the L option to the Dial application... Try show application dial at the Asterisk CLI. My guess is that this is exactly what you want. Good Luck, -- Ex Vito ___ --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] Softphone +Realtime
The first step of getting the phones to log in as their same
extensions as work is easy and works.
By definition, I guess that automatically logs out their office phones?
Has anyone tried anything like this? I would like the phones to
regrab their spot once the softphone is logged out.
Shouldn't the office phones automatically regrab their spot when they
re-register with the server? If you set the timeout to something fairly
short, it would get around this issue, but introduce another one: the
softphones will be kicked whenever the office phones re-register.
We have a number of clients doing similar things, but we've taken a slightly
different approach. For example, if we have extensions 201,202 and 203, we
create SIP accounts as follows:
201
201-home
202
202-home
203
203-home
Then, when connecting calls to those extensions in the dialplan, change
something like:
exten = _2XX,1,Dial(SIP/${EXTEN})
To:
exten = _2XX,1,Dial(SIP/${EXTEN}SIP/${EXTEN}-home)
Hopefully that'll solve the problem. Obviously you'll get lots of errors in
the logs along the lines of can't find device SIP/202-home when the
softphones aren't connected, but it shouldn't affect operation.
Regards,
Chris
--
C.M. Bagnall, Director, Minotaur I.T. Limited
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