[Asterisk-Users] Chan OH323 and overlapping digits
Hi, Perhaps there's something wrong in my config... I did some tests connecting Asterisk to an Ericsson MD110 PBX by setting up an h323 trunk. When dialling into asterisk I got some problems when the entire number is not in the setup message, i.e. I'm dialling digit by digit on the ericsson phone. Lets say I have 4001 in my extensions, and dial that from the Ericsson PBX, then the Ericsson switch is sending a h.225 setup message with a called party number 4. The oh323 channel replies with a h.225 callProceeding Message, which makes the MD110 stop sending further digits. I commented out already the s extension, so no matching pattern is found for a 4. I would have expected the channel to collect digits until a matching pattern is fount or until a timeout. Best regards Alexander ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to start to solve hardware problem?
It liiks like a motherboard problem. It's failing the initial boot. You say it booted again after two hours. Was the machine powered down during that interval. If so, I suspect you have a temperature problem. I've seen very similar problems from a defective CPU fan. Bill On 5/30/05, Ronald Wiplinger [EMAIL PROTECTED] wrote: Yesterday my * server (SuSE 9.2 pro on Athlon) just stopped, no screen, no reaction to keyboard or mouse. I get all kind of messages, or just stop during restart 1. just two lines of a code (immediately after turn on the computer) 113-AM21200-100-GB GV-RX30128D F1 2. Main Processor: AMD Athlon (tm) 64 Processor 3200+ CPUID: 0FF0 Patch ID: 0041 and BIOS line 12/14/2004-NF-CK804-6A6IFG09C-00 and it stops here 3. No SATRaid found and it stops here 4. ... No disk reserved for RAID use, RAID disable (actually it meant computer disabled, and it stops here) 5. if I come to boot Linux, than it stops somewhere, with or without kernel panic. The most I came was: Excessive leakage detected on module 0:0 volts (00) after 162 ms ProSLIC module failed leakage test. Check for short circuit DC-DC cal has a surprising direct 107 of 0xff! # Loop error (ff) # # Loop error (ff) # # Loop error (ff) # # Loop error (ff) # # Loop error (ff) # # Loop error (ff) # # Loop error (ff) # !!! DTMF_ROW_0_PEAK iREG 0 = should be 55C2 # Loop error (ff) # # Loop error (ff) # !!! DTMF_ROW_1_PEAK iREG 1 = should be 51E6 # Loop error (ff) # # Loop error (ff) # !!! DTMF_ROW_2_PEAK iREG 2 = should be 4B85 # Loop error (ff) # # Loop error (ff) # !!! DTMF_ROW_3_PEAK iREG 3 = should be 4937 # Loop error (ff) # # Loop error (ff) # !!! DTMF_ROW_1_PEAK iREG 4 = should be CPU 0: Machine Check Exception: 4 Bank 4: b20070f0f TSC 1cd4adfa6b Kernel panic - not syncing: Machine check Above error I googled as a problem in the Digium card. Chicken or egg? Digium card or a result of another problem? Since I most of the time did not come so far. Any idea? I can immagine from all the errors I saw: 1. Motherboard is broken 2. CPU is broken 3. Power supply is broken (btw a very expensive one: 200 US$) What is your expert opinion??? Where and how to start to get back the system stabled. Yes, after 2 hours it booted again, but I worry, that it will happen anytime again. bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie problem with registration of sip client
hello all, now, i want to do configuration to make sip client have extension on my asterisk.but i have a problem with registration of sip client. *CLI May 31 13:58:01 WARNING[4927]: chan_sip.c:886 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 115 (Critical Request) May 31 13:58:15 NOTICE[4927]: chan_sip.c:4585 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again May 31 13:58:21 WARNING[4927]: chan_sip.c:886 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 116 (Critical Request) May 31 13:58:35 NOTICE[4927]: chan_sip.c:4585 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again and if try the kphone from other pc to registration : May 31 14:11:10 NOTICE[4927]: chan_sip.c:9138 handle_request_register: Registration from 'mustafa sip:[EMAIL PROTECTED]' failed for '192.168.8.188' sip.conf: [general] context=default bindport=5060 bindaddr=192.168.8.125 srvlookup=yes register = mustafa:[EMAIL PROTECTED]:5060/20531604 [20531604] context=mustafa type=friend ;secret=mustafa host=dynamic defaultip=192.168.8.188 mailbox=1604 extension.conf : [mustafa] include =default exten = 20531604,1,Dial(SIP/[EMAIL PROTECTED],2Killed [EMAIL PROTECTED] asterisk]# inging exten = 20531604,3,wait(10) exten = 20531604,4,Answer exten = 20531604,5,Voicemail(s1604) exten = 20531604,6,hangup can anyone give advise and correct my configuration.. thanks.. regard, shahdan.. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] astpp database creation failed!
Hello, I'm setting up AST Post Paid application, is there anybody who set up astpp ? I followed the directions, i visited the astpp admin page in my web browser. But i couldn't setup the brands and routes etc. "Database unavailable -- please check configuration" appeared on the top of the page, so i went to "configure" section, I filled in the blanks according to my username,pass etc.. but I got "Database creation failed!" message... How can i achieve this problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 3000 dialing noise
Have you updated with the lastest firmware..
It now does an on-hook forward to asterisk
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop
Sent: Tuesday, 31 May 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sipura 3000 dialing noise
Hi all,
We have several sipura 3000's working well for outbound calls, however the
issue we have is that when calls are sent to the Sipura with
Dial(SIP/${EXTEN:[EMAIL PROTECTED]) the Sipura does a SIP answer immediately and
then proceeds with the call in band therefore sending dialing sounds back
to the caller. Other SIP gateways we have notably the Vegastream and others
do not do a SIP answer until the call is successfully connected to the
called party.
Any ideas?
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Re: [Asterisk-Users] Remote phone: Got SIP response 481 Call Leg/Transaction Does Not Exist back from
Ronald Wiplinger wrote: One of our remote user's phone reports frequently: Got SIP response 481 Call Leg/Transaction Does Not Exist back from IP What can I do ??? Turn on SIP debug, set verbose to 4, debug level to 4 and trace what happens. If we can't see that, an error message out of context will not say anything about what is going on in your server. Please give us a bit more information! Regards, /Olle --- http://www.astricon.net/europe - The Asterisk conference, Madrid June 15-17 * Full conference agenda now published ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura ATA and Asterisk No Answer Issue
Tim P wrote: I have multiple Sipura ATA 2100s attached to normal analog phones that are all configured as extensions in * When I call an extension it rings and will go to voicemail if no one answers it. When I call the same extension a second time after no answer (went to VM) the phone does not even ring but instead goes straight to voicemail. Not sure if this is a simple setting in the Sipura I missed (like a user is away setting or if there is one in * ). Has anyone else had this problem? I did a search on google and was unable to come up with anything. There are some problems with the Sipura that we currently handle in the bug tracker. This seems unrelated to those though. If you check with sip show peers when the call goes directly to voicemail - is the phone registred and reachable? /Olle --- http://www.astricon.net/europe - The Asterisk conference, Madrid June 15-17 * Full conference agenda now published ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting a peer to a dynamic ip asterisk box ???
Manjit Riat wrote: Hi, I prevoiusly has asterisk on a public static ip and had a phone from a different location registering to the asterisk box. But now we have dropped the previous connection and the current connection has a dynamic ip. Is there any way for the phone to register to now-dynamic ip addressed asterisk box (using something like dyndns.org or something). Dyndns.org seems like a good choice. Just make sure you put in the hostname in the phone configuration, not the IP address :-) /Olle --- http://www.astricon.net/europe - The Asterisk conference, Madrid June 15-17 * Full conference agenda now published ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGC on asterisk
Hi- How to configure MGCP in asterisk. I want to connect my asterisk to MGC gateway. Best Regards Ibrar Ahmed Project Manager. Comcept (Pvt) Ltd. Islamabad Pakistan www.com-cept.com [EMAIL PROTECTED] [EMAIL PROTECTED] Ph # (Off) +92-51-111784784 Ph # (Res) +92-51-2271283 Ph # (Mob) +92-3009543001 Fax # 92-51-111784785 www.com-cept.com Pick battles that are big enough to matter, small enough to win __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with asterisk+gnugk
Hi! I'm trying to build gnugk with asterisk. Asterisk is working well with chan_h323 built with needed PWlib v.1.5.2 and open H.323 v.1.12.2. But gnugk' s installing instructions says that I need latest PWlib(1.17.1) and openh323 to get gnugk work. Now, with installed pwlib and openh323 gnugk's compiling fails and I get error 1. Do you have any working solutions with asterisk and gnugk and what are needed version numbers which you use to get then work together? Thanks in advance! This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CRM integration (was RE: CallerID)
I am doing some testing using FOP (Flask Operator Panel) and so far, its going great! Been able to do callerid and also open a SugarCRM screen. All without having to install anything on the computer, just open a FOP browser screen and that's it! More later when I debug some ideas. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Adam Goryachev |Sent: Lunes, 30 de Mayo de 2005 09:18 a.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] CRM integration (was RE: CallerID) | |On Sat, 2005-05-28 at 19:19 +0100, Tom Fanning wrote: | snip | | The guy mentioned Java from within the browser. I believe that I am | right in saying that a Java applet should very well be able |to listen | for tcp connections as well as udp datagrams. Try this primer: | |http://homepages.uel.ac.uk/2795l/pages/javaapps.htm#Class%20ServerSock | et%20( | TCP%20Server%20Connections) | |Yep, thanks for replying for me... | |So, has anyone got the time + motivation to do something??? I |wish I did :( | |Regards, |Adam | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Changes on CVS HEAD
In article [EMAIL PROTECTED], Anton Krall [EMAIL PROTECTED] wrote: I just installed the latest cvs head and seems a lot of commands haven been depricated. Where can I see the changes on all cvs head versions in order to keep up with the changes needed on my side. I checked the wiki and it still shows all the old commands and no mentions about the changes. You need to subscribe to the asterisk-cvs mailing list (and asterisk-dev if you don't already). Then you will see all the changes as they happen, and can investigate in more details those of interest. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: UK DID providers
In article [EMAIL PROTECTED], Tom Fanning [EMAIL PROTECTED] wrote: Hi Can anyone provide me with a Manchester (0161) UK DID number, preferably IAX2 but SIP is ok too, that I can use for my incoming calls? Call volume will be low. The critical thing is that DTMF must be correctly passed 100% of the time, unlike Sipgate, my current (free) provider, whose DTMF detection/passing is not at all reliable, making it useless for a virtual receptionist scenario. I don't mind paying for this service (free is good though...), as long as it is reasonably less than the cost/rental of another physical BT line in to our premises. Try www.voiptalk-org - I use them with IAX2 and DTMF is fine. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 dialing noise
Yeah tried it.
Unfortunately I need this feature in reverse. I need the call to stay
on hook when going from Asterisk to Sipura. Staying onhook from Sipura
to Asterisk workd fine.
On 5/31/05, David Phelan [EMAIL PROTECTED] wrote:
Have you updated with the lastest firmware..
It now does an on-hook forward to asterisk
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop
Sent: Tuesday, 31 May 2005 3:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sipura 3000 dialing noise
Hi all,
We have several sipura 3000's working well for outbound calls, however the
issue we have is that when calls are sent to the Sipura with
Dial(SIP/${EXTEN:[EMAIL PROTECTED]) the Sipura does a SIP answer immediately
and
then proceeds with the call in band therefore sending dialing sounds back
to the caller. Other SIP gateways we have notably the Vegastream and others
do not do a SIP answer until the call is successfully connected to the
called party.
Any ideas?
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[Asterisk-Users] sox
Good day all I remember some time ago I tried recording on asterisk But it did not work because the sox app was broken and by downloading a older one it worked Now things have come and go and version change What sox version will work with asterisk 1.0.7 Thanks Altus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: cmd curl crashes asterisk:
In article [EMAIL PROTECTED],
Tim Connolly [EMAIL PROTECTED] wrote:
I recently began using the curl cmd to do an external callerid
lookup on my own customer database. I've noticed certain lookups will cause
a crash and not show anything in the messages file or the console.
It is failed lookups (or perhaps also ones that return no data) that
cause a crash. See http://bugs.digium.com/bug_view_page.php?bug_id=4389
or use CVS HEAD later than 2005/05/26
Cheers
Tony
The curl
command is connecting to an external webserver which has a oracle db
connection. The file its hitting is PHP and does a very simply lookup
showing the text like C1234 Bobs mowing service which I later cut off at
15 characters to squeeze it in setcidname(). Here is an example crash.
-- Goto (macro-getcustid,s,3)
-- Executing NoOp(Zap/2-1, Call from using .) in new stack
-- Executing Curl(Zap/2-1,
http://old-inside.theplanet.com/xmlservices/cnum_lookup.html?cid=;) in new
stack
pbx01*CLI
Disconnected from Asterisk server
[macro-getcustid]; ${DEFAULT} is my own number..i.e. no cid was given...
exten = s,1,setvar(CURL=)
exten = s,2,gotoif($[${CALLERIDNUM} = ${DEFAULT}]?9:3)
exten = s,3,noop(Call from ${CALLERIDNAME} using ${CALLERIDNUM}.)
exten =
s,4,curl('http://mywebserver-name/xmlservices/cnum_lookup.html\?cid=${CALLER
IDNUM}')
exten = s,5,setvar(CURL=${CURL:0:15})
exten = s,6,noop(Setting callerid ${CALLERIDNUM} to ${CURL})
exten = s,7,setcidname(${CURL})
exten = s,8,goto(s,10)
exten = s,9,noop(Skipping because CID = ${CALLERIDNUM})
exten = s,10,noop
I can easily avoid these crashes (I hope) by not executing the curl
command if the ${CALLERID} variable is less than 10 characters, but I
thought I would point out that CURL should not be crashing the whole system
because a URL was disliked.
Asterisk CVS-HEAD-04/14/05-15:57:59
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Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 234
Hello All I'm using asterisk 1.1.X and MFCR2 lib version 0.03pre2. when i call to E1 (connected with asterisk), chan_unicall don't detected event incoming call and show error. error messages:*CLI Warning, flexibel rate not heavily tested!Rx CAS bits 0x9 [ 1/ 0/ 0]Line unblocked -- R2 Channel 4 unblockedRx CAS bits 0x9 [ 1/ 0/ 0]Line unblocked -- R2 Channel 2 unblockedRx CAS bits 0x9 [ 1/ 0/ 0]Line unblocked -- R2 Channel 13 unblockedRx CAS bits 0x1 [ 1/ 0/ 0]Tx CAS bits 0xD [ 1/ 0/ 0]Rx CAS bits 0x9 [ 2/101/ 0]Tx CAS bits 0xD [ 2/101/ 0]ScheduleRx CAS bits 0x9 [ 1/ 0/ 0]Line unblocked -- R2 Channel 5 unblockedRx CAS bits 0x9 [ 1000 0/ 0/ 0]Line unblocked -- R2 Channel 11 unblocked Asterisk Ready.*CLI Rx CAS bits 0x9 [ 1/ 0/ 0]Line unblocked -- R2 Channel 10 unblockedRx CAS bits 0x9 [ 1/ 0/ 0]Line unblocked -- R2 Channel 2 unblockedRx CAS bits 0x1 [ 1/ 0/ 0]Line unblocked and seizedUnschedule 1Tx CAS bits 0xD [ 1/ 0/ 0] -- R2 Channel 8 unblockedRx CAS bits 0x1 [ 1/ 0/ 0]Line unblocked and seizedUnschedule 1Tx CAS bits 0xD [ 1/ 0/ 0] -- R2 Channel 9 unblockedRx CAS bits 0x1 [ 1/ 0/ 0]Line unblocked and seizedUnschedule 1Tx CAS bits 0xD [ 1/ 0/ 0] -- R2 Chan nel 6 unblockedRx CAS bits 0x1 [ 1/ 0/ 0]Line unblocked and seizedUnschedule 1Tx CAS bits 0xD [ 1/ 0/ 0] -- R2 Channel 15 unblockedRx CAS bits 0x9 [ 2/101/ 0]Tx CAS bits 0xD [ 2/101/ 0]ScheduleRx CAS bits 0x9 [ 2/101/ 0]Tx CAS bits 0xD [ 2/101/ 0]ScheduleRx CAS bits 0x9 [ 2/101/ 0]Tx CAS bits 0xD [ 2/101/ 0]ScheduleRx CAS bits 0x9 [ 2/101/ 0]Tx CAS bits 0xD [ 2/101/ 0]ScheduleRx CAS bits 0x1 [ 1/ 0/ 0]Tx CAS bits 0xD [ 1/ 0/ 0]Rx CAS bits 0x9 [ 2/101/ 0]Tx CAS bits 0xD [ 2/101/ 0]Schedule my setting zaptel.conf # Zaptel Configuration Filespan=1,0,0,cas,hdb3,crc4 cas=1-15:cas=17-31: dchan=16 alaw=1-31 loadzone=frdefaultzone=fr my setting zapata.conf ; Zapata telephony interface; Configuration file[trunkgroups]trunkgroup = 1,16spanmap = 1,1,1[channels]language=encontext=from-pstnswitchtype=nationalnsf=nonepridialplan=nationalprilocaldialplan=nationaloverlapdial=yespriindication = outofbandsignalling=pri_cperxwink=300; Atlas seems to use long (250ms) winks;usedistinctiveringdetection=yesusecallerid=yescidsignalling=bellcidstart=ringhidecallerid=nocallwaiting=yes;restrictcid=nousecallingpres=yescallwaitingcallerid=yesthreewaycalling=yestransfer=yescanpark=yescancallforward=yescallreturn=yes;mailbox=1234echocancel=yesechocancelwhenbridged=yes;echotraining=yes;echotraining=800relaxdtmf=yesrxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=no;callerid=2564286000;amaflags=default;accountcode=lss0101;adsi=yes;busydetect=yes;busycount=4;hanguponpolarityswitch;callprogress=yes;progzone=us;pulsedial=yes;faxdetect=both;faxdetect=incoming;faxdetect=outgoing;faxdetect=nomusiconhold=default;idledial=6999;[EMAIL PROTECTED];minunused=2;minidle=1;jitterbuffers=4;cadence=125,125,2000,-4000;cadence=250,250,500,1000,250,250,500,-4000; cadence= 125,125,125,125,125,-4000;cadence=1000,500,2500,-5000;crv = 1:16 my setting unicall.conf ; $Id: unicall.conf.sample,v 1.1 2005/05/28 11:17:02 steveu Exp $[channels]language=encontext=defaultusecallerid=yeshidecallerid=nocallwaitingcallerid=yesthreewaycalling=yestransfer=yescancallforward=yescallreturn=yesechocancel=yesechocancelwhenbridged=yes;relaxdtmf=yesrxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=noprotocolclass=mfcr2protocolvariant=vn,20,7protocolend=cogroup = 1channel = 1-15channel = 17-31 i'm using sangoma card, firmware V.25 and driver beta8-g.2.3.3 Please help me Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK NCFA calling
I am looking for a provider that accepts BYOD that has good rates to UK NCFA (+44 0870 ..._. If anyone knows of a provider that they use that has reliable service I would greatly appreciate hearing from it. Feel free to reply private since this isnt directly asterisk related. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ipchains for firewall, QOS howto?
I have an Asterisk PBX behind a manually-built IPCHAINS firewall machine. Can anyone tell me what I need to allow/build QOS packet rewrites through this simple NAT barrier? What do I need to pass to IPCHAINS to let QOS out to the next outside network hop? I ask this, because I have been getting intermittent jitter from my provider (TELIAX), and since it seems near-impossible to verify the source of the latency, I want to make sure I have all my Ts crossed and Is dotted before I blame something external for my issues. On the same note, what is the best way to test my connection for jitter, packet loss, etc, and still be able to determine what the potential culprit is for the problem? Thanks again, Chris Coulthurst ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UPS rating for SOHO asterisk box
Slightly OT, but I think this is of possible interest to many of you, I need to get a UPS for my asterisk box. They are rated in VA but I can't quite figure out how that converts to real life. I have a PIII-800 box with two X100P and one TDM400P plus graphics adapter, an IDE hard drive etc. Will a small 400VA box be enough for this? tia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with asterisk+gnugk
You can either download the executable version of gnugk or you can reinstall the other versions of the pwlib and openh323 as they are only needed during the compile. RG, Gentian - Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, May 31, 2005 8:14 AM Subject: [Asterisk-Users] Problem with asterisk+gnugk Hi! I'm trying to build gnugk with asterisk. Asterisk is working well with chan_h323 built with needed PWlib v.1.5.2 and open H.323 v.1.12.2. But gnugk' s installing instructions says that I need latest PWlib(1.17.1) and openh323 to get gnugk work. Now, with installed pwlib and openh323 gnugk's compiling fails and I get error 1. Do you have any working solutions with asterisk and gnugk and what are needed version numbers which you use to get then work together? Thanks in advance! This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] handytone 486
Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mmkn olmayabilir. Mesaji alan kisi, mesajin gnderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkrler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk install error ...
Title: Asterisk install error ... Hi; It is my first time to use asterisk I have TDM400 wildcard and 4 FXO Modules when I install asterisk an error occurred Chen_zap.c 2772 : error : zt_event_dtmfdigit undeclared Can any body help why this error .. Thanks; Ghassan M. Lama' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk install error ...
Title: Asterisk install error ... Hi; Thanks for replay; I have used the latest CVS and the stable version . I am installing the software on Fedora core 2 Kerenl 2.6 I do have zaptel instaled and configured Regrds; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk: HelpDesk / CRM type of Application in Asterisk
hi, I am new to asterisk I have a client who wants a help desk type of application. the asterisk tool kit seems to fit the bill nicely. is there anything already implemented that is available or is all the asterisk implementations custom? attributes of the project include a sequence of events as follows: we need two call paths for different scenarios call flow path 'A' call in, get CID and produce a screen pop, if no match on our database, ask for name (screener stage I) agent #1 fills in / updates a form then pass call to an extension (agent #2, screener stage II) w/ screen pop on local PC, agent #2 has the option of passing that db record directly or creating/completing a second form. this would either have multiple monitors (i.e. a teleprompter) or pass it on to the final destination. record the call at each stage call flow path 'B' call in, get CID and produce a screen pop, if no match on our database, ask for name (agent #1 screener stage I) lookup agent criteria matching the caller's needs, send screen pop to a remote PC (even over the Internet) (stage II) then transfer the call to the remote PC via VoIP if local, PSTN if not local (stage III) record the call at each stage any help would be appreciated... thanks, dave cantera -- The eyes of the Lord roam over the whole earth, to encourage those who are devoted to Him wholeheartedly. II Chronicles 16:9 NAB ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Uniden UIP1868 - any sightings or users?
I've been looking out for the Uniden UIP1868 for a while now, but I haven't seen it anwhere that I'm used to buying things from. According to froogle, a couple of places (that I've never heard of) have a small number in stock (small = 10 in this case). I'm doubly suspicious because even uniden's own online store doesn't have them available yet, not to mention reputable places like voipsupply.com. Uniden's product support doesn't list it either. Has anybody seen one in the flesh?And more importantly, are they actually out yet? And if not, any ideas when? -- Peter Wemm - [EMAIL PROTECTED]; [EMAIL PROTECTED]; [EMAIL PROTECTED] All of this is for nothing if we don't go to the stars - JMS/B5 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS rating for SOHO asterisk box
Hi, Let me try to answer this one. Assuming your P3-800 is using a 300watt power supply, then in a full load condition, convert to VA, it will be 300/0.6=500VA. So, it is greater than your small 400VA box. So, you need a bigger ups. Of course, if your power usage is actually much lower than 300watt which in most cases it is, then your 400va box may be ok, but then you are taking a risk. Another thing to consider regarding the ups is the runtime, depending on the hours and minutes you want the ups to supply power to your asterisk box, you may need to add more batteries to the ups. cheers Wilson Pickett wrote: Slightly OT, but I think this is of possible interest to many of you, I need to get a UPS for my asterisk box. They are rated in VA but I can't quite figure out how that converts to real life. I have a PIII-800 box with two X100P and one TDM400P plus graphics adapter, an IDE hard drive etc. Will a small 400VA box be enough for this? tia ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting a peer to a dynamic ip asterisk box ???
Dyndns.org seems like a good choice. Just make sure you put in the hostname in the phone configuration, not the IP address :-) Also, when the ip changes, users will usually need to reboot their phones. I added a mail alert that sends a heads up to users and also some stuff to reprovision the IAXy when the ip changes (as it does NOT do DNS). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Connecting a peer to a dynamic ip asterisk box ???
GS phones don't need to reboot. On Tue, 2005-05-31 at 10:58 +0200, Wilson Pickett wrote: Dyndns.org seems like a good choice. Just make sure you put in the hostname in the phone configuration, not the IP address :-) Also, when the ip changes, users will usually need to reboot their phones. I added a mail alert that sends a heads up to users and also some stuff to reprovision the IAXy when the ip changes (as it does NOT do DNS). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS rating for SOHO asterisk box
Another thing to consider regarding the ups is the runtime, depending on the hours and minutes you want the ups to supply power to your asterisk box, you may need to add more batteries to the ups. Regarding this, I have done this hack yesterday: - Remove the battery from an existing UPS - Rewire the UPS onto biggest car lead acid battery (12v) you can find. Et voila! Bigger capacity. Put the batteries in parrallel and you do get monstruous UPS capacity... the only trouble with it is that re-charging the batteries may take some time. -- Ykoz Un Max - La VoIP en pré-payé! Essayez gratuitement - 5 crédits offerts. --- http://ykoz.net/voip/max --- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS rating for SOHO asterisk box
Assuming your P3-800 is using a 300watt power supply, then in a full load condition, convert to VA, it will be 300/0.6=500VA. So, it is Thanks for that info. Where does the /0.6 come from? I've always wondered about VA which looks like VoltAmps. There are 400, 500 and 600VA models. The asterisk box is alone on the UPS so I guess a 500 should be the best investment. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS rating for SOHO asterisk box
On Tue, 2005-05-31 at 13:22 +0400, Jean-Michel Hiver wrote: Another thing to consider regarding the ups is the runtime, depending on the hours and minutes you want the ups to supply power to your asterisk box, you may need to add more batteries to the ups. Regarding this, I have done this hack yesterday: - Remove the battery from an existing UPS - Rewire the UPS onto biggest car lead acid battery (12v) you can find. Et voila! Bigger capacity. Put the batteries in parrallel and you do get monstruous UPS capacity... the only trouble with it is that re-charging the batteries may take some time. And place the battery in a well ventilated environment :) You could give it a boost with a car charger. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to configure Inter7's Asterisk Fax with Postfix
Tzafrir, We need to send an email with the fax number for astfax to fax. eg: From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: ... attach fax image How to configure postfix to understand and deliver this? I've tried putting this line in /etc/aliases : fax:|/var/mail/ast_fax/ast_fax /var/mail/ast_fax/ast_fax.call It don't seems to works. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extension context question
I have a very simple question .
I have 2 internal extension 301 and 300 sip phone . I want to these extesioncan call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal.
How can I do that ?
[x1]exten = 300,1,Dial(SIP/300)
include = pstnlocal
[x2]exten = 301,1,Dial(SIP/301)
include =international
[pstnlocal]
exten = _9xxx,1,Dial(Zap/g1/${EXTEN})
[international]
exten = _900.,1,Dial(Zap/g1/${EXTEN})
So it is good but in this case I cann t call the local phone .And if I include context x1 in x2 and x2 in x1 the ext 300 will be able to call international no.
Can anyone help me ?
Thanks.
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[Asterisk-Users] How does ISDN really work?
I'm trying to setup DATA calls with Dial(Zap/g1d/12345678), but with PRI DEBUG SPAN 1 on, it seems to connect a regular SPEECH call. I'm using 1.0.6. Is this feature broken in stable release? There seems to be support in the source, but it doesn't work. Does the Telco set what each PRI channel support? Like DATA or SPEECH etc.. Do I have to specify in zapata.conf or zaptel.conf that the channels are DATA capabale? Please help! This is driving me crazy soon. :) -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS rating for SOHO asterisk box
Normally, the power factor is taken as 0.6, thus to convert watt to va, just divid the wattage by 0.6 to get the va rating. cheer Wilson Pickett wrote: Assuming your P3-800 is using a 300watt power supply, then in a full load condition, convert to VA, it will be 300/0.6=500VA. So, it is Thanks for that info. Where does the /0.6 come from? I've always wondered about VA which looks like VoltAmps. There are 400, 500 and 600VA models. The asterisk box is alone on the UPS so I guess a 500 should be the best investment. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How does ISDN really work?
Hi, you will need app_settransfercapability to make this work properly. This is part of CVS-HEAD. I have backported it for the asterisk stable version of bristuff (see www.junghanns.net/asterisk/) and also fixed some bugs in Asterisk that will make ISDN data calls unreliable (or in some cases impossible). On the * CLI do a show application settransfercapability to find out the correct arguments. best regards Klaus -- Klaus-Peter Junghanns Am Dienstag, den 31.05.2005, 11:54 +0200 schrieb Daniel Nystrm: I'm trying to setup DATA calls with Dial(Zap/g1d/12345678), but with PRI DEBUG SPAN 1 on, it seems to connect a regular SPEECH call. I'm using 1.0.6. Is this feature broken in stable release? There seems to be support in the source, but it doesn't work. Does the Telco set what each PRI channel support? Like DATA or SPEECH etc.. Do I have to specify in zapata.conf or zaptel.conf that the channels are DATA capabale? Please help! This is driving me crazy soon. :) -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to configure Inter7's Asterisk Fax with Postfix
On Tue, May 31, 2005 at 05:38:55PM +0800, Eddie wrote: Tzafrir, We need to send an email with the fax number for astfax to fax. eg: From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: ... attach fax image How to configure postfix to understand and deliver this? I've tried putting this line in /etc/aliases : fax:|/var/mail/ast_fax/ast_fax /var/mail/ast_fax/ast_fax.call It don't seems to works. This is basically a postfix question and not an asterisk question. What you should do is probably define an alternative transport, fax, in /etc/postfix/master.cf . See the following example for defining the transport maildrop: http://www.postfix.org/MAILDROP_README.html This will keep the local method for actual mail delivery (e.g: cron jobs to root). You should be able to define such exceptions in the transports file. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk compailation Error Chan_zap.c
Title: Asterisk compailation Error Chan_zap.c Hi; It is my first time installing an asterisk PBX system I do have a TDM400 wildcard with 4 FXO moduls on a PC with 3.0GHZ HT CPU and INTEL 915 moatherboard Fedora C2 Linux as O.S. and I have the latest CVS astreisk , Zaptel and Libpri downloaded the zaptel drivers installation and configuration seems to be fine and the libpri but when I tried to compile and install the asterisk software the following error occurred : Chan_zap.c 2772 : error : Zt_event_DTMFDIGIT undeclared Can any body help why this error .. Thanks; Ghassan M. Lama' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UPS rating for SOHO asterisk box
Another thing to consider regarding the ups is the runtime, depending on the hours and minutes you want the ups to supply power to your asterisk box, you may need to add more batteries to the ups. No worry there, since the modems (upstairs) will be unpowered as well. Although the asterisk box recovers pretty well from a short power cut, it is risky to allow these to happen at all. This is why I want the UPS, not for autonomy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] handytone 486
Betl Gzlkolu wrote: Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... On my 486 I can't dial out on the FXO port, it's just a lifeline. There are rumours that there is an update or a new version that can do this. The SIPURA 3000 supports this and work with Asterisk. /Olle Astricon - the Asterisk User's conference - Madrid June 15-17 http://www.astricon.net/europe/ - Register today! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension context question
asterisk asterisk wrote: I have a very simple question . I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal. How can I do that ? You read the samples and the guides that are available on the Voip-info.org wiki, in the Asterisk Documentation project and other places. After reading those, configuring Asterisk to do this will be a piece of cake :-) This page is a good start: http://www.voip-info.org/tiki-index.php?page=Asterisk Find the guides and articles under the Articles heading. I especially like John Todd's articles on the ONLAMP web site. Good luck! /O Astricon - the Asterisk User's conference - Madrid June 15-17 http://www.astricon.net/europe/ - Register today! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension context question
asterisk asterisk wrote:
I have a very simple question .
I have 2 internal extension 301 and 300 sip phone . I want to these
extesion can call each other, and ext 300 can call outside to pstn,
and ext 301 to call internatonal.
How can I do that ?
include pstnlocal at either [x2] or [international]
Remember, that international starts with 00 while local never has 00 -
right?
bye
Ronald
[x1]
exten = 300,1,Dial(SIP/300)
include = pstnlocal
[x2]
exten = 301,1,Dial(SIP/301)
include =international
include = pstnlocal
[pstnlocal]
exten = _9xxx,1,Dial(Zap/g1/${EXTEN})
[international]
exten = _900.,1,Dial(Zap/g1/${EXTEN})
include = pstnlocal
So it is good but in this case I cann t call the local phone .And if I
include context x1 in x2 and x2 in x1 the ext 300 will be able to call
international no.
Can anyone help me ?
Thanks.
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[Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working
Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear the other party (to be more precise I can hear first two secs then nothing). So it must be the incoming RTP is blocked on Linksys. Here I think STUN server enters the game and give some help? I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? -- David Hajek http://hajek.net/blog ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto-generated incoming calls X100P
Hi, I am struggling with a problem where calls are being created on ZAP channel, but no call exists. I have 2 X100P cards 1st = BT, 2nd Telewest, with no problems on BT line. I have carried out various test on signalling types, Kewlstart, Loopstart, Groundstart and EM. Only Kewlstart and Loopstart appear to work, but Kewlstart generates incoming calls every 20 - 30 sec, where as Loopstart can run for upto 5 min without an auto generated incoming call. BT line if running Kewlstart ok. Have tried both cards to ensure it is not hardware issue. Checked physical wiring, which all seems good. Can anyone suggest further changes/tests? Thanks in advance. Robert Brown I can post zaptel and zapata files if that will help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ztdummy usage
Dear All, I have installed Asterisk everything is OK until I tried to configure meeting room, configuration was simple enough when I try I get a message that it's not a valid meeting room, Now I don't have a Zaptel device on my machine, so I found that you will have to use ztdummy to make a dummy zaptel device on your machine and this is because of timing issues. My question is ztdummy can only be done when making asterisk or is ther a way to do it after post installation? I am using by the way freebsd 5.3, built Asterisk from the ports successfully. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working
I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear the other party (to be more precise I can hear first two secs then nothing). So it must be the incoming RTP is blocked on Linksys. Here I think STUN server enters the game and give some help? I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? Add canreinvite=no to the sipura def's in sip.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ztdummy usage
You can build it alone. Then try to 'modprobe zaptel' and then 'modprobe ztdummy' If you do this without errors it must work. For more info read the wiki. RG, Gentian - Original Message - From: Mohamed A. Gombolaty To: asterisk-users@lists.digium.com Sent: Tuesday, May 31, 2005 12:22 PM Subject: [Asterisk-Users] Ztdummy usage Dear All, I have installed Asterisk everything is OK until I tried to configure meeting room, configuration was simple enough when I try I get a message that it's not a valid meeting room, Now I don't have a Zaptel device on my machine, so I found that you will have to use ztdummy to make a dummy zaptel device on your machine and this is because of timing issues. My question is ztdummy can only be done when making asterisk or is ther a way to do it after post installation? I am using by the way freebsd 5.3, built Asterisk from the ports successfully. -- Thx MAG ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automatic Codec change for different communication channels!?
Hi, I am looking for a way to let * choose the voice codec relying to the used communication channel. Example I am using a Polycom 500 which supports G729 and G.711. When I am doing internal calls (with my LAN) or calls over the PSTN (ISDN) I want to use the G.711 codec because there is enough bandwith. When I am doing inter asterisk calls (over my WAN to another * server) I want to use G.729. Is there a way how i can achieve this? Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compailation Error Chan_zap.c
Dear Ghassan, I never used fedora but in the link below you will find a step by step installation for fedora platform check it out and see if you are missing anything. http://www.voip-info.org/wiki-Asterisk+Linux+Fedora Thx MAG Ghassan Lama wrote: Hi; It is my first time installing an asterisk PBX system ... I do have a TDM400 wildcard with 4 FXO moduls on a PC with 3.0GHZ HT CPU and INTEL 915 moatherboard ... Fedora C2 Linux as O.S. and I have the latest CVS astreisk , Zaptel and Libpri downloaded the zaptel drivers installation and configuration seems to be fine and the libpri but when I tried to compile and install the asterisk software the following error occurred : Chan_zap.c 2772 : error : "Zt_event_DTMFDIGIT" undeclared Can any body help why this error .. Thanks; Ghassan M. Lama' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000 behind NAT issue, Vonage is working
I have canreinvite=no already, below is my sip.conf entry. [1360] username=1360 callerid=Phone 1 1360 secret=mysec1 host=dynamic auth=md5 qualify=1000 dtmfmode=rfc2833 context=from-sip-unrestricted mailbox=1360 type=friend disallow=all allow=ilbc allow=g729 allow=gsm allow=g726 nat=yes canreinvite=no - David Hajek http://hajek.net/blog Rich Adamson wrote: I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear the other party (to be more precise I can hear first two secs then nothing). So it must be the incoming RTP is blocked on Linksys. Here I think STUN server enters the game and give some help? I have installed Vovida STUN server and point Sipura to use it. But no luck, I still can't hear the other party. I've ended up with having Linksys to forward all ports to my Sipura (DMZ host) which works. What is interesting is that when I'm using Vonage service (Cisco ATA) it works just fine without touching the Linksys. How come they can get through it? Any hints? Add canreinvite=no to the sipura def's in sip.conf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] monitoring oh323 calls
Hello list, I put together a quick note about how to see oh323 calls while they are handled by your * box. http://www.oinko.net/astrecipes/index.php?n=89 The article is just a draft with usage examples; I'd love to hear your comments and updates if there is something I got wrong. Thanks l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] monitoring oh323 calls
FYI If using oh323 v0.6.5 then the oh323 show info has been replaced by the command oh323 show channels Thanks Giles -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of lenz Sent: 31 May 2005 12:51 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] monitoring oh323 calls Hello list, I put together a quick note about how to see oh323 calls while they are handled by your * box. http://www.oinko.net/astrecipes/index.php?n=89 The article is just a draft with usage examples; I'd love to hear your comments and updates if there is something I got wrong. Thanks l. -- Assum est, versa et manduca. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension context question
Yes Pstn local start with 9 and pstn international starts with 00 .That is ok.
I can make call form 300 to pstn local, and from ext 301 to pstn international, that is ok .But in this exemple I can not call formext 300 to 301 and form 301 to 300.
It is possibile to have2 diferent group of extension, with diferent tyipe of permision (like here x1 pstn local , and x2 pstn international) but extension can call each other regardless form the group. It does not matther in which group is the pone I can call any phone form group x1 or x2.
But if I want to call outside it depends on permision of group in wich I am (x1 or x2).
Ronald Wiplinger [EMAIL PROTECTED] wrote:
asterisk asterisk wrote: I have a very simple question . I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal. How can I do that ? include pstnlocal at either [x2] or [international]Remember, that international starts with 00 while local never has 00 - right?byeRonald [x1] exten = 300,1,Dial(SIP/300) include = pstnlocal [x2] exten = 301,1,Dial(SIP/301) include =internationalinclude = pstnlocal [pstnlocal] exten = _9xxx,1,Dial(Zap/g1/${EXTEN}) [international] exten
=
_900.,1,Dial(Zap/g1/${EXTEN}) include = pstnlocal So it is good but in this case I cann t call the local phone .And if I include context x1 in x2 and x2 in x1 the ext 300 will be able to call international no. Can anyone help me ? Thanks. Do You Yahoo!? Yahoo! Small Business - Try our new Resources site! ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Ronald Wiplinger (CEO of ELMIT)http://www.elmit.com +886 (0) 939--77-55-16 or FWD 511208- I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.orgPS: Spam prevention!Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [Asterisk-Users] Automatic Codec change for different communication channels!?
you can try use variable preffered_codec in dial command (if you now the prefixes/dial numbers, for which to use eg. g729)... PJ Kib Eki wrote: Hi, I am looking for a way to let * choose the voice codec relying to the used communication channel. Example I am using a Polycom 500 which supports G729 and G.711. When I am doing internal calls (with my LAN) or calls over the PSTN (ISDN) I want to use the G.711 codec because there is enough bandwith. When I am doing inter asterisk calls (over my WAN to another * server) I want to use G.729. Is there a way how i can achieve this? Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Soft Video phone for Asterisk usage
Nardis Dome wrote: try eyeBeam, it works fine for me... [] type=friend secret= auth=md5 callerid=myCallerId canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 Thanks, I bought eyeBeam for two computers on the LAN for testing, but I get with above settings on both screens: Remote party does not support video What do I miss? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 3000 dialing noise
Eric
A completly off topic response (and not even a response in that I'm
asking you a question - sorry)
you say that you have several 3000 devices and you show your dial string as :
Dial(SIP/${EXTEN:[EMAIL PROTECTED])
Is the sipura1 section referencing a single sipura or the group of
several. The only reason that I ask if it is the latter - how are
you grouping them.
Thanks for you response if i figure the answer to your question I
will post back.
David
On 31/05/05, Eric Bishop [EMAIL PROTECTED] wrote:
Hi all,
We have several sipura 3000's working well for outbound calls, however
the issue we have is that when calls are sent to the Sipura with
Dial(SIP/${EXTEN:[EMAIL PROTECTED]) the Sipura does a SIP answer immediately
and then proceeds with the call in band therefore sending dialing
sounds back to the caller. Other SIP gateways we have notably the
Vegastream and others do not do a SIP answer until the call is
successfully connected to the called party.
Any ideas?
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Re: [Asterisk-Users] Automatic Codec change for different communication channels!?
could you please give more information concerning this setting? Pavel Jezek wrote: you can try use variable preffered_codec in dial command (if you now the prefixes/dial numbers, for which to use eg. g729)... PJ Kib Eki wrote: Hi, I am looking for a way to let * choose the voice codec relying to the used communication channel. Example I am using a Polycom 500 which supports G729 and G.711. When I am doing internal calls (with my LAN) or calls over the PSTN (ISDN) I want to use the G.711 codec because there is enough bandwith. When I am doing inter asterisk calls (over my WAN to another * server) I want to use G.729. Is there a way how i can achieve this? Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Soft Video phone for Asterisk usage
Hi, did you enable the right video-codecs in eyeBeam? settings-media-video-Advanced-Codecs --- Ronald Wiplinger [EMAIL PROTECTED] wrote: Nardis Dome wrote: try eyeBeam, it works fine for me... [] type=friend secret= auth=md5 callerid=myCallerId canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 Thanks, I bought eyeBeam for two computers on the LAN for testing, but I get with above settings on both screens: Remote party does not support video What do I miss? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 'beeps' while recording..?
How would one go about having asterisk insert a faint 'beep' once every ten seconds or so whilst a call is being recorded? I can't see any flags in the Monitor appliction for doing so. This is a legal requirement in many jurisdictions when calls are being recorded (as an alternative to an announcement at the start). Ben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tools for effectively manage Asterisk
Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards standardizing installed modules, functionalities, tools etc. The wall we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to us); it also suffers from its prerequisites: apache, mysql, php... too much things that should not go in a pbx We tried IPSwitchboard, but it seems only good as a monitor, not as a configuration tool (are we correct or are we missing something?) At this point we are thinking that we better abandon the idea of GUI tools and that we must go on the road of vi editing of .conf files. We would like to understand what other people are using for asterisk management, and to get some suggestion from the community. Any suggestion is welcome Francesco Pellegrini ++ | Frame Srl | | Via Antonio Cantore 62/10 | | 16149 Genova | | Tel. +39 010 8680570| | Fax. +39 010 6591413 | | Cell. +348 2237798 | ++ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Soft Video phone for Asterisk usage
Nardis Dome wrote: Hi, did you enable the right video-codecs in eyeBeam? settings-media-video-Advanced-Codecs I have here 1. H.263++QCIF 128 2. H.263+ 3. Basic H.263 and in asterisk allow = 'ulaw;alaw;speex;gsm;h263;h263p' --- Ronald Wiplinger [EMAIL PROTECTED] wrote: Nardis Dome wrote: try eyeBeam, it works fine for me... [] type=friend secret= auth=md5 callerid=myCallerId canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 Thanks, I bought eyeBeam for two computers on the LAN for testing, but I get with above settings on both screens: Remote party does not support video What do I miss? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Chan OH323 and overlapping digits
There is nothing wrong with your config, it is just unimplemented functionality. Michael. Alexander Topolanek wrote: Hi, Perhaps there's something wrong in my config... I did some tests connecting Asterisk to an Ericsson MD110 PBX by setting up an h323 trunk. When dialling into asterisk I got some problems when the entire number is not in the setup message, i.e. I'm dialling digit by digit on the ericsson phone. Lets say I have 4001 in my extensions, and dial that from the Ericsson PBX, then the Ericsson switch is sending a h.225 setup message with a called party number 4. The oh323 channel replies with a h.225 callProceeding Message, which makes the MD110 stop sending further digits. I commented out already the s extension, so no matching pattern is found for a 4. I would have expected the channel to collect digits until a matching pattern is fount or until a timeout. Best regards Alexander ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tools for effectively manage Asterisk
On Tue, 2005-05-31 at 15:08 +0200, [EMAIL PROTECTED] wrote: Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards standardizing installed modules, functionalities, tools etc. The wall we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to us); it also suffers from its prerequisites: apache, mysql, php... too much things that should not go in a pbx We tried IPSwitchboard, but it seems only good as a monitor, not as a configuration tool (are we correct or are we missing something?) At this point we are thinking that we better abandon the idea of GUI tools and that we must go on the road of vi editing of .conf files. We would like to understand what other people are using for asterisk management, and to get some suggestion from the community. Vi is certainly a simple and stable solution. One idea that works well for us, is to put similar static type configs together. For example: put all the remote sip entries in one config and all the local sip entries in another, than include them both in your sip.conf file. It would be interesting if vi would highlight the * entries. -- respectfully, Joseph === -= ** = ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automatic Codec change for different communication channels!?
This is off the top of my head - never tested For the end user device (ie polycom in your case) your sip settings would be something like [5000] username=5000 SNIP deny=all allow=ulaw allow=alaw allow=G729 which would give you both Then if you in the Trunk set the following [Trunkroute] username=asterisk2 SNIP deny=all allow=G729 Actually thinking some more, this might not work, as your asterisk box may transcode it, although it might not - but even if my logic is flawed here, it might inspire? David On 31/05/05, Kib Eki [EMAIL PROTECTED] wrote: could you please give more information concerning this setting? Pavel Jezek wrote: you can try use variable preffered_codec in dial command (if you now the prefixes/dial numbers, for which to use eg. g729)... PJ Kib Eki wrote: Hi, I am looking for a way to let * choose the voice codec relying to the used communication channel. Example I am using a Polycom 500 which supports G729 and G.711. When I am doing internal calls (with my LAN) or calls over the PSTN (ISDN) I want to use the G.711 codec because there is enough bandwith. When I am doing inter asterisk calls (over my WAN to another * server) I want to use G.729. Is there a way how i can achieve this? Kib ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with another Asterisk
Hi, I'm a newbie on Asterisk and I'd like to know if it's possible to connect two or more asterisk together. In fact, I'd like install and connect some asterisk together. Thanks for advance, Cyril _ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tools for effectively manage Asterisk
[EMAIL PROTECTED] wrote: We tried AMP, very powerful but incomplete (CAPI is very important to us); The 1.10.008 version of AMP supports Custom Trunks. Text from the AMP tooltip: -begin- Define the custom Dial String. Include the token $OUTNUM$ wherever the number to dial should go. examples: CAPI/:b$OUTNUM$,30,r H323/[EMAIL PROTECTED] OH323/[EMAIL PROTECTED]: vpb/1-1/$OUTNUM$ -end- Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tools for effectively manage Asterisk
[EMAIL PROTECTED] wrote: Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards standardizing installed modules, functionalities, tools etc. The wall we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to us); it also suffers from its prerequisites: apache, mysql, php... too much things that should not go in a pbx We tried IPSwitchboard, but it seems only good as a monitor, not as a configuration tool (are we correct or are we missing something?) At this point we are thinking that we better abandon the idea of GUI tools and that we must go on the road of vi editing of .conf files. We would like to understand what other people are using for asterisk management, and to get some suggestion from the community. Any suggestion is welcome We are working on finalizing a production release of our PhoneCALL product, a GPL php/smarty configuration GUI for Asterisk: http://www.vecsector.com/phonecall I feel there is nothing wrong with having a web-based configuration utility, if set up correctly. Look at the WRT54G Linksys router, plus other countless devices that use an embedded browser for configurations. It can save a lot of time on training new employees, and syntax issues when starting out. Our goal is to have a GUI that is just as flexible as writing configurations by hand, but not having to write it by hand. ;-) PhoneCALL is not production ready yet, we are on 2.5-RC4 - but within a week or so, we plan to have a very nice/clean stable version that is production ready. We don't have CAPI support built-in yet, but open for any help anyone would like to lend. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 MWI
I've google'd this to death, is there a simple way to make MWI work from * for my Cisco phone ??? Examples ??? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with another Asterisk
Yes, via IAX Adam Collard General Manager, ER Wireless (800) 757-5669 x4861 (810) 496-0161 Fax (517) 242-1800 Cell Nextel DC 131*256784*19 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cyril SIMON Sent: Tuesday, May 31, 2005 1:29 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with another Asterisk Hi, I'm a newbie on Asterisk and I'd like to know if it's possible to connect two or more asterisk together. In fact, I'd like install and connect some asterisk together. Thanks for advance, Cyril _ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with another Asterisk
On 31/05/05, cyril SIMON [EMAIL PROTECTED] wrote: Hi, I'm a newbie on Asterisk and I'd like to know if it's possible to connect two or more asterisk together. In fact, I'd like install and connect some asterisk together. As usual, Google and the Wiki are there for your convenience. http://www.voip-info.org/wiki-Asterisk+Connect+2+servers Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk sip register with no username and password.
I am connecting to a local nortel PBX. The person setting up the PBX did not define a user name and password (dont ask why). So I presume my register command can leave off the username and password and look like: [sip.conf] register localpbx/5551212 I presume that is good enough then so calls coming in to 5551212 would get routed over to my asterisk box? Thanks Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3000 Analog Line No Answer, No Audio
Problem 1 - Outgoing: I am able to call out of the * box using the analog line attached to the sipura 3000 but when the person being called answers there is no audio from either end. * registers that the call was answered but passes no audio. Problem 2 - Incoming: When calling into the 3000 attached to * it never seems to pickup the line. The phones don't ring on the asterisk side. I used the below writeup to create the extensions for Line 1 and PSTN in the 3000 as well as creating the Trunk, extensions and DID routes in *. Can someone give me an idea of where to start with the troubleshooting here? I am kind of lost as to where to begin. Thanks! ### In AMP add an extension (e.g. 200) to correspond to Line 1 on the SPA, ensure that port is 5060 and context is from-internal. Add a second extension (e.g. 280) for PSTN Line on SPA, ensure that port is 5061 and set context to from-pstn (disable voicemail directory on this extension). In Trunks add a Sip trunk and copy the Outgoing block as follows (just leave Incoming as it is - do not delete the any defaults, but you do not need to change them either).: Trunk name sipura1 context=from-pstn fromuser=280 (or whatever extension you used) host=IP address of you SPA (needs to be fixed IP) port=5061 secret=your password type=peer username=280 (or whatever extension you used) Inbound User context sipura1-in Leave defaults in Inbound box and leave Register String blank. In DID Routes, add DID with a unique string (I used S followed by the PSTN number that the SPA is attached to - e.g. S12345678 Set an outbound route using the new sipura1 trunk. On the SPA 3000: Do the following configuration in admin login, advanced mode: In Line 1, make sure SIP port is 5060, proxy points to your * Box, no outbound proxy. Fill out subscriber info with settings above e.g. User ID = 200, Password =***, Display Name =***. Set your preffered codec. In PSTN Line, ensure SIP Port = 5061 proxy = Asterisk Box IP, no outbound proxy. Fill out subscriber info with Display Name =, User ID = 280 (or whatever you used), Password =. Set preferred codec. It is vital that you Set Dial Plan 8 to (S0:S12345678) (or whatever string you used for the DID route in Asterisk). Ensure that both VoIP-To-PSTN Gateway Enable and PSTN-To-VoIP Gateway Enable are set to yes. Set PSTN Caller Default DP to 8. If you want incoming calls to all be sent to * then set PSTN Ring Thru Line 1 to no. Set PSTN Answer Delay to the number of seconds that you want the phone to ring for before sending it to your * box. Leave other settings on the SPA at factory defaults until you really know what you're doing and want to fine-tune things. Lastly, make sure you plug into the line jack into the SPA and not the jack marked phone! I know this seems obvious, but I've missed this simple step before! The only kink with inbound using the settings posted is that you can't have it ring to a phone plugged into the Sipura's phone port. You can still call out, and the system will still pick up the call if you have auto attendant recieve the calls. But, if you set the inbound calls to ring extension 200, your calls will just go directly to voicemail. That aside, you can have any other phone on the system ring for inbound calls directly, or set a ring group. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] astpp database creation failed!
The most common problem is that the web server does not have permission to write to the config files in /var/lib/astpp. Try changing their ownership to be the same as the web server owner. Darren Wiebe [EMAIL PROTECTED] Erdem HAKI wrote: Hello, /I'm setting up AST Post Paid application, is there anybody who set up astpp ?/ I followed the directions, i visited the astpp admin page in my web browser. But i couldn't setup the brands and routes etc. Database unavailable -- please check configuration appeared on the top of the page, so i went to configure section, I filled in the blanks according to my username,pass etc.. but I got Database creation failed! message... How can i achieve this problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Soft Video phone for Asterisk usage
Ronald Wiplinger wrote: Nardis Dome wrote: Hi, did you enable the right video-codecs in eyeBeam? settings-media-video-Advanced-Codecs I have here 1. H.263++QCIF 128 2. H.263+ 3. Basic H.263 Try 261? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recommendations for good quality stylish * compatible phones
I already have the Cisco 7960 and am thinking of getting a Polycom 501 or 600. Anyone got these? Quality? One phone I really like is the Mitel 5240 but unfortunately there is no SIP image available for them. Anyone know of a phone similar in looks/functions that works with * ? Thanks in advance Phil. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_sccp / wiki
The chan_sccp page at http://www.voip-info.org/tiki-index.php?page=chan_sccp2 has been updated. See the bottom of the page. Thanks. Comments welcome. -- respectfully, Joseph === -= ** = ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: R: R: [Asterisk-Users] AT-320 + supervised transfer
Hi Gavin, I installed the cvs Asterisk CVS-D2005.05.28.22.00.00-05/31/05-14:25:23 and i added this rowd in the features.conf [featuremap] blindxfer = #1; Blind transfer disconnect = *0 ; Disconnect automon = *1 ; One Touch Record atxfer = *22 ; Attended transfer But...how atxfer work ? Thanks -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Gavin Hamill Inviato: lunedì 30 maggio 2005 18.40 A: asterisk-users@lists.digium.com Oggetto: Re: R: R: [Asterisk-Users] AT-320 + supervised transfer On Monday 30 May 2005 17:22, Giordano Grandis wrote: The procedure that will do asterisk is very nice ;) but whe it was available ? Asterisk's atxfer support is only in CVS. Currently is there any way to emprove the transfer? I tryied the scenario that u suggest me but it doesn't work :| and i don't why. You *must* be using a new firmware for the phone. Download 1.43 from http://www.aredfox.com/edownloadssip.htm (the AT-320 needs PA186S code) Here my sip.conf for the phone, can u say me if there is somethingh wrong ? Looks fine to me.. I think is ok, maybe i have some problem on phone settings.Can I see your exmple phone setting ? They're at work so I can't see the config right now... but they're just the defaults with the DTMF changed to RFC2833 and the NTP server set... Try resetting to defaults using the procedure at http://www.voip-info.org/wiki-ATCOM+AT-320 Cheers, Gavin. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tools for effectively manage Asterisk
Dustin Wildes wrote: I feel there is nothing wrong with having a web-based configuration utility, if set up correctly. Look at the WRT54G Linksys router, plus other countless devices that use an embedded browser for configurations. Just a nitpick, if I may. They have embedded http servers, not browsers. But I'm sure that's what you meant. Having said that, I agree that putting streamlined apache/php on an * box isn't going to cause grief. Heck, I'm breaking lots of rules, and haven't running into problems (yet). I run _everything_ on my Athlon 3000+/1GB Gentoo machine. Apache, postfix, named, mysql, courier-imap, firebird / avg tcp server, nagios, samba, X/Gnome, and vncserver/Gnome! I even (gasp) play some games on it. I'm sure that slows down some of the server functions, but I haven't noticed any problems (yet). I'm hoping to get my own dedicated server box soon to offload all the non-client stuff, but until then, it all goes on this one machine. Yes, this is a home setup, but with ties to work functions. - Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with another Asterisk
Has anyone seen a situation where, upon connecting two asterisk servers together with IAX registration, outgoing/incoming calls that route through both servers are choppy and jittery? I don't have this problem when I call out to teliax (my ITSP) directly, but if I try to make the call through the 'remote' asterisk server downtown, it gets bad. If I register my SIP phone here at home to the server downtown directly and make the call, the problem goes away again! CPU load is low, and the cable internet pipe is free and wide open with no appreciable latency. I've tried every jitterbuffer config I could think of. Any suggestions on where to find some probable causes? Chris -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Collard Sent: Tuesday, May 31, 2005 6:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk with another Asterisk Yes, via IAX Adam Collard General Manager, ER Wireless (800) 757-5669 x4861 (810) 496-0161 Fax (517) 242-1800 Cell Nextel DC 131*256784*19 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cyril SIMON Sent: Tuesday, May 31, 2005 1:29 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk with another Asterisk Hi, I'm a newbie on Asterisk and I'd like to know if it's possible to connect two or more asterisk together. In fact, I'd like install and connect some asterisk together. Thanks for advance, Cyril _ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: astpp database creation failed...please help...
so what should astpp db be exactly, where can i find its name? what should i write there? Thanks again.. The Database field should contain the name of the astpp db, something along the lines of astpp is what I would put in there. Here is a fixed version of the script. It did not post properly to the wiki: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Error in Zapata Config?
Chris Mason (Lists) wrote: When I reload the config, I see this error in the CLI. However, I don't see what I have done wrong: == Parsing '/etc/asterisk/zapata.conf': Found May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 1, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 2, FXO Kewlstart signalling May 30 16:38:42 WARNING[12630]: chan_zap.c:10088 setup_zap: Ignoring signalling -- Reconfigured channel 3, FXS Kewlstart signalling -- Reconfigured channel 4, FXS Kewlstart signalling Sorry, where does it say error? All I see is a warning (because you have reloaded and those settings are ignored on a reload). -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ANNOUNCEMENTt: GPL Asterisk Billing Software
Sorry, no support for rates with time limits yet. You can file a bug @ http://www.aleph-com.net/astpp/ if you wish. Darren Wiebe [EMAIL PROTECTED] Erik Versaevel - Infopact Netwerkdiensten BV wrote: What happens if the rate changes mid call? IE, call starts @ 18.30 and lasts till 19.15 Rate changes @1900 to off-peak. Darren Wiebe wrote: Partially. I have not finished the script that will limit the calls depending on the money available. Darren Wiebe [EMAIL PROTECTED] VoIP Newbie wrote: Does it support pre-paid billing? On 5/30/05, Darren Wiebe [EMAIL PROTECTED] wrote: El Flynn wrote: Darren Wiebe wrote: Good Day, I'm finally getting around to officially announcing ASTPP. For the last 6 months, I've been working on converting ASTCC into a decent billing package for asterisk. The link in the original email opens a page that says Download the latest version of the code from http://www.aleph-com.net/astpp.html Has anyone else been able to download this code? I can't seem to find a link on their site to the code itself, and the astpp.html page brings up a Not Found... Sorry, I missed that old link. I just got everything moved onto the wiki on Friday night. Please download the code off of the cvs server. I'm getting close to ready to release version 1.0 and then I will post a copy on the website. At present, I believe the only show stopping bug is in the AgileBill integration. That will be fixed shortly. Darren Wiebe [EMAIL PROTECTED] Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 MWI
[EMAIL PROTECTED] wrote: I've google'd this to death, is there a simple way to make MWI work from * for my Cisco phone ??? Examples ??? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Should be easy. Just add 'mailbox=extension' in your sip.conf under the entry. Example: [1003] type=friend username=1003 secret=mysecret nat=no host=dynamic mailbox=1003 does the MWI for Cisco phones. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Tools for effectively manage Asterisk
What is it you feel is missing in AMP? Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 31 May 2005 9:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Tools for effectively manage Asterisk Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards standardizing installed modules, functionalities, tools etc. The wall we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to us); it also suffers from its prerequisites: apache, mysql, php... too much things that should not go in a pbx We tried IPSwitchboard, but it seems only good as a monitor, not as a configuration tool (are we correct or are we missing something?) At this point we are thinking that we better abandon the idea of GUI tools and that we must go on the road of vi editing of .conf files. We would like to understand what other people are using for asterisk management, and to get some suggestion from the community. Any suggestion is welcome Francesco Pellegrini ++ | Frame Srl | | Via Antonio Cantore 62/10 | | 16149 Genova | | Tel. +39 010 8680570| | Fax. +39 010 6591413 | | Cell. +348 2237798 | ++ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Invalid login/password with AreskiCC V2
Did you install php-pgsql? Check if the register_global is On in php.ini file (reload apache) Regards, A. On 5/31/05, Alexandre Charles [EMAIL PROTECTED] wrote: Hi Everybody, I have tried to make AreskiCCV2 work on RH9.0 but it does not work. More precisely, I have followed the guide as well as the installation instructions but I always get an Invalid login/password error when i try to login using the web interface. The login/password provided do match in all the configuration files. Any clues? Any comments on the applications? Any alternative to the application? Thanks in advance, AC Lèche-vitrine ou lèche-écran ? Yahoo! Magasinage. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Cisco 7960 MWI
[EMAIL PROTECTED] uttered the following thing: I've google'd this to death, is there a simple way to make MWI work from * for my Cisco phone ??? Examples ??? Message waiting? Sure... If you're using SIP, then it will work as long as you have the right 'mailbox=' line in your sip peer config. I get a nice bright red indicator on my 7960 plus stuttered dialtone. For SCCP I can't help though. BB ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk on Soekris
So, I'm wondering does anyone have real-life comparisons on the failure rate of a PC compared to the failure rate of some of these options?? Obviously, an embedded PC or something that is designed such as a Sokeris is made to last a *long* time, but in my experience, a Tier 1 PC (older Compaq, HP, *not* consumer) PC fares well. I use old Tier 1 PC's for utility jobs like small firewalls or FTP servers or hell even homebrew SAN's and the like, and they just keep chugging. I've never seen a power supply die on a Deskpro, and I've been using them for 10 years. They seem immune to the stupid minor problems that bring clones to a halt, like dust in the fans. I'd never use a clone in an an application where the life expectancy is greater than a year. I sleep well at night knowing that all of those old PC's will be quietly running and doing their jobs just fine the next day. Also, Tier 1 PC's typically are well documented, you can still get drivers for them, and the design is consitient and *made* for business applications. For example, every Deskpro ever made allows you to run it headless, there's an option for it right in the BIOS. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Connecting a peer to a dynamic ip asterisk b ox ???
Some ISP's provide a static hostname on a dynamic host, which you can use to your advantage. Ask them if it is possible. For example, up where I am an extremely large ISP is Telus Communications. They require you to register the host's MAC address with an online tool and when you do, the tool shows you a static hostname which you can use to address the host regardless of the IP address. -Original Message- From: Manjit Riat [mailto:[EMAIL PROTECTED] Sent: Monday, May 30, 2005 5:48 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Connecting a peer to a dynamic ip asterisk box ??? Hi, I prevoiusly has asterisk on a public static ip and had a phone from a different location registering to the asterisk box. But now we have dropped the previous connection and the current connection has a dynamic ip. Is there any way for the phone to register to now-dynamic ip addressed asterisk box (using something like dyndns.org or something). Thanx ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tools for effectively manage Asterisk
Jason, thanks a lot for the info. Is there any way to separate AMP stuff from asterisk, in other words to have AMP, apache and so on on a different pbx than asterisk? Tia brgds Francesco Pellegrini Frame srl [EMAIL PROTECTED] Jason Becker [EMAIL PROTECTED] systems.caTo Sent by: Asterisk Users Mailing List - asterisk-users-bo Non-Commercial Discussion [EMAIL PROTECTED] asterisk-users@lists.digium.com m.com cc Subject 31/05/2005 15.28 Re: [Asterisk-Users] Tools for effectively manage Asterisk Please respond to Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] ists.digium.com [EMAIL PROTECTED] wrote: We tried AMP, very powerful but incomplete (CAPI is very important to us); The 1.10.008 version of AMP supports Custom Trunks. Text from the AMP tooltip: -begin- Define the custom Dial String. Include the token $OUTNUM$ wherever the number to dial should go. examples: CAPI/:b$OUTNUM$,30,r H323/[EMAIL PROTECTED] OH323/[EMAIL PROTECTED]: vpb/1-1/$OUTNUM$ -end- Regards, -- Jason Becker Director CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Soft Video phone for Asterisk usage
in your sip.conf: [general] videosupport=yes ; in your eyeBeam settings- try to enable all the h.263 codec. hope it helps... --- Ronald Wiplinger [EMAIL PROTECTED] wrote: Nardis Dome wrote: Hi, did you enable the right video-codecs in eyeBeam? settings-media-video-Advanced-Codecs I have here 1. H.263++QCIF 128 2. H.263+ 3. Basic H.263 and in asterisk allow = 'ulaw;alaw;speex;gsm;h263;h263p' --- Ronald Wiplinger [EMAIL PROTECTED] wrote: Nardis Dome wrote: try eyeBeam, it works fine for me... [] type=friend secret= auth=md5 callerid=myCallerId canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 Thanks, I bought eyeBeam for two computers on the LAN for testing, but I get with above settings on both screens: Remote party does not support video What do I miss? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: R: R: [Asterisk-Users] AT-320 + supervised transfer
On Tuesday 31 May 2005 14:41, Giordano Grandis wrote: Hi Gavin, But...how atxfer work ? Ehm, just the way I explained yesterday :) Just make sure you include the 't' option to the Dial application, in the same way you need for the old-style '#' blind-transfer to function. gdh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Soft Video phone for Asterisk usage
Just got it working with eyebeam: in sip.conf under general: videosupport=yes allow=h261 allow=h263 shouldn't need per phone config. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Tuesday, May 31, 2005 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Soft Video phone for Asterisk usage Ronald Wiplinger wrote: Nardis Dome wrote: Hi, did you enable the right video-codecs in eyeBeam? settings-media-video-Advanced-Codecs I have here 1. H.263++QCIF 128 2. H.263+ 3. Basic H.263 Try 261? -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with another Asterisk
Has anyone seen a situation where, upon connecting two asterisk servers together with IAX registration, outgoing/incoming calls that route through both servers are choppy and jittery? I don't have this problem when I call out to teliax (my ITSP) directly, but if I try to make the call through the I found this problem minimised when I used the same codec end-to-end. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco 7960 MWI
Works for me, make sure you're not sending the voicemail to an e-mail account, no point in setting the MWI in that instance. Here's my Voicemail.conf... format=wav serveremail=asterisk attach=yes skipms=3000 maxsilence=10 silencethreshold=128 maxlogins=3 sendvoicemail=yes [zonemessages] eastern=America/New_York|'vm-received' Q 'digits/at' IMp central=America/Chicago|'vm-received' Q 'digits/at' IMp central24=America/Chicago|'vm-received' q 'digits/at' H 'digits/hundred' M 'hours' [default] 1234 = 4242,Example Mailbox,[EMAIL PROTECTED] 2002002001 = 1234,Andrew Herdman Andrew -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, May 31, 2005 9:34 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cisco 7960 MWI I've google'd this to death, is there a simple way to make MWI work from * for my Cisco phone ??? Examples ??? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS rating for SOHO asterisk box
I have a PIII-800 box with two X100P and one TDM400P plus graphics adapter, an IDE hard drive etc. Will a small 400VA box be enough for this? It's tricky sizing UPS'es to be bang on the money. The rule-of-thumb calculation for VA is watts/.6 . So, for a 200 watt power supply / .6 is 333 VA. Tricky part is, a 200 watt PSU is max power, which your PC will not draw all the time. You want to be bang on the money, so you have to determine the watts of every individual component like this: (numbers out of my ass) HDD 35W Mobo CPU 80W Video 10W NIC 5W TDM400 20W TOTAL 160W = 266 VA In my experience, oversizing your UPS gives you a comfortable margin, and these days, pricing between, say, a 400 to 600 VA is minor, and you get added runtime as a bonus. But if you want a decent runtime on your box, and it's of the 200W PSU variety, the 400VA will probably be OK given what you have stated. hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] # Transfers
I am currently running stable, CVS-v1-0-05/25/05-12:07:15, with Polycom SIP phones, running 1.4.1. Too many of our transfers using the Transfer end up with zombie channels after a REFER. As such, I implemented # transfers, and all is well. Sort of. I have a reproducible issue. Take a call from a queue. Press #, and it'll transfer just fine. Now, take a call from the queue. Put them on hold for a couple seconds. Pick them back up and press #. They hear a beautiful, short, DTMF tone, nothing more. Is this a bug, or did I miss something in the configurations? Has anyone else had this problem? As far as the transfers, I found a message at http://lists.digium.com/pipermail/asterisk-users/2004-September/062080.h tml but there were no more messages in that thread. The other zombie channel transfer questions didn't seem to fit the problem, but I may be wrong. Any suggestions would be greatly appreciated. Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UPS rating for SOHO asterisk box
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Tuesday, May 31, 2005 5:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] UPS rating for SOHO asterisk box [...] Regarding this, I have done this hack yesterday: - Remove the battery from an existing UPS - Rewire the UPS onto biggest car lead acid battery (12v) you can find. Et voila! Bigger capacity. Put the batteries in parrallel and you do get monstruous UPS capacity... the only trouble with it is that re-charging the batteries may take some time. [...] Congratulationsyou've just given this part-time small town fire marshal and 14-year fire service veteran nightmares. Kidsdo NOT try this at home. The inverters in small UPSes are not designed to deal with runtimes that exceed the batteries in them. If you run this setup well past the time it was designed to run (by adding 3, 4, or more times that battery capacity it was ever designed to have) that chances of a catastrophic inverter failure (meaning flash, boom, fire) are very real and very likely. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uniden UIP1868 - any sightings or users?
Peter - I speak with the folks at Uniden regularly, the UIP1868 currently has an ETA of late June, although I expect it might be into July before these are widely available. Unless there are eval units floating around, to my knowledge, these are not available in the channel yet. Cory Andrews Senior Partner VOIPSupply.com + V: 800.398.VOIP X22 F: 716.630.1548 E: [EMAIL PROTECTED] Peter Wemm wrote: I've been looking out for the Uniden UIP1868 for a while now, but I haven't seen it anwhere that I'm used to buying things from. According to froogle, a couple of places (that I've never heard of) have a small number in stock (small = 10 in this case). I'm doubly suspicious because even uniden's own online store doesn't have them available yet, not to mention reputable places like voipsupply.com. Uniden's product support doesn't list it either. Has anybody seen one in the flesh?And more importantly, are they actually out yet? And if not, any ideas when? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Built-In Transfer Questions
I've read the Wiki on using asterisk's built-in transfer options (#8 and #6). They work fine but how does one cancle an attended transfer? Example: I have person on phone, I hit #6 to being att-transfer. I enter Sally's extension. I let it ring for a few seconds. Sally never picks up but her voicemail does. How do I hangup her voicemail and resume the previous call? The example on the wiki assumes the transferee picks up the phone. :/ -Matthew -- Matthew Boehm, IT DirectorCypress Telecommunications [EMAIL PROTECTED] 3838 N. Sam Houston Parkway E #400 T: 832-200-8640 x3044 Houston, TX 77032 My girlfriend was recently diagnosed with multiple personality disorder; When she called yesterday, my CallerID box exploded. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
