[Asterisk-Users] Agent Callback, how to see wath queue is calling the agent?
This is the problem:
two Queues
Agent logged in as agentcallback and member of the two queues.
When a call come in the queue, asterisk call the extension provided
by the agentcallbacklogin.
The need is in the extension to have a variable with the queue id.
something like:
exten = _6XXX,1,Noop(AGENT ${EXTEN:1})
exten = _6XXX,n,UserEvent(FOP_Popup|URL:
/cmgr/api/popup?e=${EXTEN:1}cid=${CALLERIDNUM}q=${QUEUE})
exten = _6XXX,n,DIAL(ZAP/g2/1234567${EXTEN:1},30)
exten = _6XXX,n,Hangup
Any idea?
Thnks.
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Re: [Asterisk-Users] Is NuFone Really Dead?
On 5/24/06, Andy Jefferson [EMAIL PROTECTED] wrote: Went to their site today. Site claims they are still in biz. What is the story? What really happened to Nufone anyway? The word dead isn't too accurate. If you pronounced dead and were buried while in a temporary coma, you'd see that. or not :) Nufone outgoing service has not been interrupted at all, at least not for me. They have a new agreement with a different provider, I assume, because I ordered a new 800 number from them a few days ago and it works great. The old ones are being processed, I'm not sure what is happening behind the scenes. Support is important when you have a problem, but asking for support when they've clearly said they won't be able to answer questions or discuss this issue is more than optimistic. Yes, many DIDs are down, and if you depend on them for your business, you (no one in particular, anyone here) should definitely have backup numbers. This said, you can't failover to new numbers in the current situation AFAIK, you'd have to tell people the new number. This works fine for us, but I know it won't for most people. Nufone is not currently dead although I don't know whatthe future will bring for them. I wish them the best, they're still better than many other providers and I have tested at least 15. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Modules for X100P
Can anybody recommend a reseller in Europe (Netherlands) for modules for the X100P (FXO and FXS modules)? Cost, support are important. Also, what is a reasonable price for an X100P in Europe? Is there a difference in price between OEM and Boxed versions? Thanks, Pieter ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] # key
On 5/25/06, Akpome Akpoguma [EMAIL PROTECTED] wrote: I was actually running record() application, when I pressed the # key to interrupt the recording, it just doesnt stop This can depend on features.conf, the codec used, the phone used, the digitmap of the phone if there is one and several other things. You would need to describe what your equipment is. Do numeric keys work (do DTMF tones)? Does the * key work? Search for features.conf on google and look at the wiki, search this list for features.conf. Somewhere in all of that you will find helpful information. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No sound when the call is diverted
Hi Guys,
I'm having sound problems when diverting a call using [EMAIL PROTECTED] 1.5. I
am using the following configuration in extensions_custom.conf,
extensions_additional.conf and extensions.conf
[custom-Sales]
exten = s,1,SetVar(DivertNumber=02)
exten = s,2,Dial(SIP/116, 15)
exten = s,3,Goto(outrt-010-outside3,9${DivertNumber},1)
(i have replaced the diverted phone number with above)
[outrt-010-outside3] it's the context to make outbound calls via SIP trunk
The custom-Sales context is used in the following ext-did context for
incoming calls,
[ext-did]
exten = 02,1,SetVar(FROM_DID=02);
exten = 02,2,Goto(custom-Sales,s,1) ;
(i have replaced the called DID number with above)
So when ringing 02, after 15 seconds the call is successfully
diverted to 02
however when the call is answered there is not any sound on any end. Can any
one that has this
working please point me on the right direction I will appreciate it. I'm not
too sure what
would be affecting the sound on the call as it is diverted.
See below for relevant debug output from the console.
-- Executing SetVar(SIP/02-a1a7, FROM_DID=02) in new
stack
-- Executing Goto(SIP/02-a1a7, custom-Sales|s|1) in new
stack
-- Goto (custom-Sales,s,1)
-- Executing SetVar(SIP/-a1a7, DivertNumber=02) in
new stack
-- Executing Dial(SIP/02-a1a7, SIP/116| 15) in new stack
-- Called 116
-- SIP/116-ca11 is ringing
.
.
.
-- Executing SetVar(SIP/02-e487, DIAL_NUMBER=02) in
new stack
-- Executing SetVar(SIP/02-e487, DIAL_TRUNK=11) in new stack
-- Executing AGI(SIP/02-e487, fixlocalprefix) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing SetVar(SIP/02-e487, OUTNUM=02) in new
stack
-- Executing Cut(SIP/02-e487, custom=OUT_11|:|1) in new
stack
-- Executing GotoIf(SIP/02-e487, 0?20) in new stack
-- Executing NoOp(SIP/02-e487, 02) in new stack
-- Executing Dial(SIP/02-e487, SIP/sales/02) in new
stack
-- Called sales/02
-- SIP/sales-7d0b is making progress passing it to SIP/02-e487
-- SIP/sales-7d0b answered SIP/02-e487
-- Attempting native bridge of SIP/0282058347-e487 and SIP/sales-7d0b
asterisk*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Format
202.177.222.24 02 01f672b7696 00103/0 g729
202.177.222.24 02 447542a4000 00101/31350 g729
4 active SIP channel(s)
(I changed the numbers to and in the debug output as well)
Thanks in advance,
Paul
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Re: [Asterisk-Users] Asterisk codec negotiation patch
Certainly not since it's not working properly yet. Kevin P. Fleming wrote: Erick Perez wrote: does anybody knows if this patch made it into Asterisk Business Edition? http://bugs.digium.com/view.php?id=4825 ABE never includes any features that are not in open source Asterisk, except for things that cannot be done via an open source license. No patches from Mantis will ever be in ABE before they are merged in open source Asterisk. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Erik Versaevel ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Changing firmware revs did not help, so that left the LAN. I looked long and hard at the LAN and it was basically narrowed down to the switches. In this smaller install, several cheapo Dlink ($30) switches de-aggregate a Cisco Catalyst switch. What I noticed was that any phone plugged direcly into the Catalyst did *not* lock up or reboot. Any phone plugged into the crap switches experienced the lockup. So now we are down to the cheap switches themselves. We are nuking the Dlink switches and replacing them with 3com workgroup switches, same as what we use in the large install to good effect, and I fully expect the problem to dissapear. It's unfortunate that Snoms have a propensity to freak out in certain environments but I don't think it would preclude me from using Snom in the future. As long as one is aware of this issue, it should be easy enough to work around. Thanks for your input! Previously I was using Nortel 10/100 switches, I replaced them some weeks ago with 3C16479 gbit switches. The phones are connected directly to the gbit switches. By coincidence I dit notice on one phone that in a split second a message appeared 'Ethernet cable disconnected'. Because I have cable unplug set to ignore the conversation was not interrupted and the conversation could continue. But that still doesn't solve the occasional lockup. One phone was giving me *lots* more reboots than others but that was due to it running firmware 6.0.4 without having the ramdisk converted to jffs. Apparently the firmware didn't like that at all or just runs out of memory and decides to reboot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
I looked long and hard at the LAN and it was basically narrowed down to the switches. In this smaller install, several cheapo Dlink ($30) switches de-aggregate a Cisco Catalyst switch. What I noticed was that any phone plugged direcly into the Catalyst did *not* lock up or reboot. Any phone plugged into the crap switches experienced the lockup. So now we are down to the cheap switches themselves. We are nuking the Dlink switches and replacing them with 3com workgroup switches, same as what we use in the large install to good effect, and I fully expect the problem to dissapear. We had the same problems with some cheap LevelOne Switches. The Snoms rebooted during a call, calls dropped etc. Replacing the switches was the solution. Guido ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Not able to make any calls
Hi All,
I have registered abhijit for SIP in asterisk Server.
I am able to register my softphone (SJPhone) to the server using the
name abhijit.
But whenever I try to make any calls I am gettinh the following error
message:-
*CLI
-- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120
May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: Cannot
find extension context 'from-internal'
May 26 07:35:23 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: Cannot
find extension context 'from-internal'
my extension.conf is :-
[globals]
VM_PREFIX = *
RINGTIMER = 15
REGTIME = 7:55-17:05
REGDAYS = mon-fri
RECORDEXTEN =
PARKNOTIFY = SIP/200
OUT_2 = IAX2/fwd
OUT_1 = ZAP/g0
OUTPREFIX_2 =
OUTMAXCHANS_2 = 1
OUTCID_2 = mithunafila672648
OPERATOR =
NULL =
IN_OVERRIDE = forcereghours
INCOMING = GRP-1
FAX_RX_EMAIL = [EMAIL PROTECTED]
FAX_RX = system
FAX =
Eabhijit = SIP
E9002 = SIP
E9001 = SIP
E8002 = SIP
E8001 = SIP
DIRECTORY_OPTS =
DIRECTORY = last
DIAL_OUT_1 = 9
DIAL_OUT = 9
DIAL_OPTIONS = tr
DIALOUTIDS = 1/2/
CALLFILENAME =
AFTER_INCOMING =
[ext-did]
include = ext-did-custom
exten = 672648,1,SetVar(FROM_DID=672648) ;
exten = 672648,2,Goto(ext-group,1,1) ;
[ext-group]
include = ext-group-custom
exten = 1,1,Macro(rg-group,30,,200-201);
[ext-local]
include = ext-local-custom
exten = 8001,1,Macro(exten-vm,[EMAIL PROTECTED],8001)
exten = ${VM_PREFIX}8001,1,Macro(vm,8001)
exten = 8002,1,Macro(exten-vm,[EMAIL PROTECTED],8002)
exten = ${VM_PREFIX}8002,1,Macro(vm,8002)
exten = 9001,1,Macro(exten-vm,[EMAIL PROTECTED],9001)
exten = ${VM_PREFIX}9001,1,Macro(vm,9001)
exten = 9002,1,Macro(exten-vm,[EMAIL PROTECTED],9002)
exten = ${VM_PREFIX}9002,1,Macro(vm,9002)
exten = abhijit,1,Macro(exten-vm,[EMAIL PROTECTED],abhijit)
exten = ${VM_PREFIX}abhijit,1,Macro(vm,abhijit)
[outbound-allroutes]
include = outbound-allroutes-custom
include = outrt-001-9_outside
include = outrt-002-outgoingFWD
[outbound-trunks]
include = outbound-trunks-custom
exten = _${DIAL_OUT_1}.,1,Macro(dialout,1,${EXTEN})
[outrt-001-9_outside]
include = outrt-001-9_outside-custom
exten = _9.,1,Macro(dialout-trunk,1,${EXTEN:1})
exten = _9.,2,Macro(outisbusy) ; No available circuits
[outrt-002-outgoingFWD]
include = outrt-002-outgoingFWD-custom
exten = 393,1,Macro(dialout-trunk,2,${EXTEN},)
exten = 393,2,Macro(outisbusy) ; No available circuits
[internal]
exten = 100,1,Dial(SIP/abhijit)
exten = abhijit,1,Echo()
Can anyone please help ..
Abhijit.
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RE: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
On Fri, 2006-05-26 at 10:11 +0200, Remco Barende wrote: Thanks for your input! Previously I was using Nortel 10/100 switches, I replaced them some weeks ago with 3C16479 gbit switches. The phones are connected directly to the gbit switches. By coincidence I dit notice on one phone that in a split second a message appeared 'Ethernet cable disconnected'. Because I have cable unplug set to ignore the conversation was not interrupted and the conversation could continue. But that still doesn't solve the occasional lockup. One phone was giving me *lots* more reboots than others but that was due to it running firmware 6.0.4 without having the ramdisk converted to jffs. Apparently the firmware didn't like that at all or just runs out of memory and decides to reboot. Looks like you're getting somewhere now. That was my real complaint xyz sucks helps no one. As I said in my reply I've never had such problems with SNOM, perhaps it's because I've always used decent switches. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP call problem
Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only supports g723.1 pass-thru! allow=g729 The extensions.conf: [interne] ignorepat = 9 exten = 9,1,Dial(SIP/[EMAIL PROTECTED]) exten = 9,2,Hungup So when i take the phone connected to FXO (ZAP/3) and tape 9 asterisk initiate the sip call but does not succeed. *CLI sip show registry HostUsername Refresh State IP_SIP_PROVIDER:5060 testcomclien45 Registered *CLI *CLI -- Starting simple switch on 'Zap/3-1' May 26 09:49:02 DEBUG[3242]: chan_zap.c:4242 zt_read: DTMF digit: 9 on Zap/3-1 May 26 09:49:02 DEBUG[3242]: chan_zap.c:1384 zt_enable_ec: No echocancellation requested -- Executing Dial(Zap/3-1, SIP/[EMAIL PROTECTED]) in new stack May 26 09:49:02 DEBUG[3242]: chan_sip.c:1309 create_addr: Setting NAT on RTP to 0 May 26 09:49:02 DEBUG[3242]: chan_sip.c:1487 sip_call: Outgoing Call for 151 May 26 09:49:02 DEBUG[3242]: chan_sip.c:1592 update_user_counter: 151 is not a local user -- Called [EMAIL PROTECTED] May 26 09:49:03 DEBUG[3227]: chan_sip.c:822 __sip_ack: Acked pending invite 102 May 26 09:49:03 DEBUG[3227]: chan_sip.c:840 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found May 26 09:49:03 DEBUG[3227]: chan_sip.c:872 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 103: Found May 26 09:49:05 DEBUG[3227]: chan_sip.c:872 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 103: Found May 26 09:49:05 DEBUG[3227]: chan_sip.c:2780 process_sdp: Oooh, we need to change our formats since our peer supports only 0x1 (g723) and not 0x4 (ulaw) May 26 09:49:05 NOTICE[3227]: channel.c:1757 ast_set_read_format: Unable to find a path from g723 to ulaw May 26 09:49:05 NOTICE[3227]: channel.c:1724 ast_set_write_format: Unable to find a path from ulaw to g723 -- SIP/SIP_PROVIDER-77e1 is making progress passing it to Zap/3-1 May 26 09:49:05 DEBUG[3242]: chan_zap.c:4479 zt_indicate: Received AST_CONTROL_PROGRESS on Zap/3-1 May 26 09:49:05 WARNING[3242]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) May 26 09:49:05 WARNING[3242]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) May 26 09:49:05 WARNING[3242]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) May 26 09:49:05 DEBUG[3227]: chan_sip.c:872 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 103: Found -- SIP/SIP_PROVIDER-77e1 is ringing May 26 09:49:05 WARNING[3242]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) May 26 09:49:05 WARNING[3242]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) May 26 09:49:05 WARNING[3242]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) May 26 09:49:05 WARNING[3242]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) May 26 09:49:06 WARNING[3242]: chan_sip.c:1829 sip_write: Asked to transmit frame type 4, while native formats is 1 (read/write = 4/4) May 26 09:49:06 WARNING[3242]: chan_zap.c:4367 zt_write: Cannot handle frames in 1 format May 26 09:49:06 WARNING[3242]: app_dial.c:369 wait_for_answer: Unable to forward frame May 26 09:49:06 DEBUG[3242]: chan_sip.c:1716 sip_hangup: update_user_counter(151) - decrement outUse counter May 26 09:49:06 DEBUG[3242]: chan_sip.c:1592 update_user_counter: 151 is not a local user May 26 09:49:06 DEBUG[3242]: app_dial.c:1052 dial_exec: Exiting with DIALSTATUS=CANCEL. == Spawn extension (default, 9, 1) exited non-zero on 'Zap/3-1' May 26 09:49:06 DEBUG[3242]: chan_zap.c:2164 zt_hangup: Hangup: channel: 3 index = 0, normal = 21, callwait = -1, thirdcall = -1 May 26 09:49:06 DEBUG[3242]: chan_zap.c:2577 zt_setoption: Set option TDD MODE, value: OFF(0) on Zap/3-1 May 26 09:49:06 DEBUG[3242]: chan_zap.c:1352 update_conf: Updated conferencing on 3, with 0 conference users -- Hungup 'Zap/3-1' May 26 09:49:06 DEBUG[3227]: chan_sip.c:822 __sip_ack: Acked pending invite 103 May 26 09:49:06 DEBUG[3227]: chan_sip.c:840 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 103: Found May 26 09:49:06 DEBUG[3227]: chan_sip.c:840 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Request 103: Found May 26 09:49:06
[Asterisk-Users] Asterisk.NET authentication problem
Hi Im very new to Asterisk and this is my first posting to this mailing list. I got a [EMAIL PROTECTED] V2.8 working, and now Im trying to use Asterisk.NET (http://sourceforge.net/projects/asterisk-dotnet) to get call events to my C# program. Asterisk.NET comes with a sample program called Asterisk.NET.Test and it uses the following default constants for login: const int ASTERISK_PORT = 5038; const string ASTERISK_HOST = 10.34.9.206; const string ASTERISK_LOGINNAME = admin; const string ASTERISK_LOGINPWD = amp111; However, when the application tries to login using these constants I get an Authentication Failed message. In /var/log/asterisk/full log: May 26 08:06:33 DEBUG[28367] manager.c: Manager received command 'Login' May 26 08:06:33 VERBOSE[28367] logger.c: == Parsing '/etc/asterisk/manager.conf': May 26 08:06:33 VERBOSE[28367] logger.c: == Parsing '/etc/asterisk/manager.conf': Found May 26 08:06:33 VERBOSE[28367] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': May 26 08:06:33 VERBOSE[28367] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Found May 26 08:06:33 WARNING[28367] config.c: Unknown directive 'permit=192.168.1.0/255.255.255.0' at line 18 of manager_custom.conf May 26 08:06:33 DEBUG[28367] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer May 26 08:06:33 DEBUG[28367] acl.c: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer May 26 08:06:33 DEBUG[28367] acl.c: # Testing 10.34.9.135 with 0.0.0.0 May 26 08:06:33 DEBUG[28367] acl.c: # Testing 10.34.9.135 with 127.0.0.0 May 26 08:06:33 NOTICE[28367] manager.c: 10.34.9.135 failed to pass IP ACL as 'admin' May 26 08:06:33 DEBUG[27570] manager.c: Manager received command 'Command' May 26 08:06:33 DEBUG[27570] manager.c: Manager received command 'Command' May 26 08:06:34 VERBOSE[28367] logger.c: == Connect attempt from '10.34.9.135' unable to authenticate May 26 08:06:46 DEBUG[2855] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found May 26 08:06:59 DEBUG[2855] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' May 26 08:06:59 DEBUG[2855] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Does anybody know if the login name and password used is correct, or what I can try to fix this problem? Werner Werner Terreblanche CONTROL INSTRUMENTS TELEMATICS Tel: +27 21 880 5500 / 5686 (direct) Fax: +27 21 880 1756 Mobile: +27 82 3037669 [EMAIL PROTECTED] www.ci-omnibridge.com Read our disclaimer at http://www.ci-omnibridge.com/contact/disclaimer.aspx ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
On Fri, 26 May 2006, Dave Cotton wrote: On Fri, 2006-05-26 at 10:11 +0200, Remco Barende wrote: Thanks for your input! Previously I was using Nortel 10/100 switches, I replaced them some weeks ago with 3C16479 gbit switches. The phones are connected directly to the gbit switches. By coincidence I dit notice on one phone that in a split second a message appeared 'Ethernet cable disconnected'. Because I have cable unplug set to ignore the conversation was not interrupted and the conversation could continue. But that still doesn't solve the occasional lockup. Looks like you're getting somewhere now. That was my real complaint xyz sucks helps no one. As I said in my reply I've never had such problems with SNOM, perhaps it's because I've always used decent switches. You mean that 3Com switches are not to be regarded as decent switches? At least Snom could have put some remark then that you need a certain brand of switches. If 3Com is not good enough for the phones I would have bought different phones. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk.NET authentication problem
My guess would be to check your manager.conf[admin]secret = amp111deny=0.0.0.0/0.0.0.0permit=10.0.0.1/255.255.255.0read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,userThe line permit=10.0.0.1/255.255.255.0 should be adjust to your network configurations.Hope it helps! Best regards,Marco MoutaPs.Please let me know if it worked.On 5/26/06, Werner Terreblanche [EMAIL PROTECTED] wrote: Hi I'm very new to Asterisk and this is my first posting to this mailing list. I got a [EMAIL PROTECTED] V2.8 working, and now I'm trying to use Asterisk.NET (http://sourceforge.net/projects/asterisk-dotnet ) to get call events to my C# program. Asterisk.NET comes with a sample program called Asterisk.NET.Test and it uses the following default constants for login: const int ASTERISK_PORT = 5038; const string ASTERISK_HOST = 10.34.9.206; const string ASTERISK_LOGINNAME = admin ; const string ASTERISK_LOGINPWD = amp111; However, when the application tries to login using these constants I get an "Authentication Failed" message. In /var/log/asterisk/full log: May 26 08:06:33 DEBUG[28367] manager.c: Manager received command 'Login' May 26 08:06:33 VERBOSE[28367] logger.c: == Parsing '/etc/asterisk/manager.conf': May 26 08:06:33 VERBOSE[28367] logger.c: == Parsing '/etc/asterisk/manager.conf': Found May 26 08:06:33 VERBOSE[28367] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': May 26 08:06:33 VERBOSE[28367] logger.c: == Parsing '/etc/asterisk/manager_custom.conf': Found May 26 08:06:33 WARNING[28367] config.c: Unknown directive 'permit=192.168.1.0/255.255.255.0' at line 18 of manager_custom.conf May 26 08:06:33 DEBUG[28367] acl.c: 0.0.0.0/0.0.0.0/0.0.0.0 appended to acl for peer May 26 08:06:33 DEBUG[28367] acl.c: 127.0.0.1/255.255.255.0/255.255.255.0 appended to acl for peer May 26 08:06:33 DEBUG[28367] acl.c: # Testing 10.34.9.135 with 0.0.0.0 May 26 08:06:33 DEBUG[28367] acl.c: # Testing 10.34.9.135 with 127.0.0.0 May 26 08:06:33 NOTICE[28367] manager.c: 10.34.9.135 failed to pass IP ACL as 'admin' May 26 08:06:33 DEBUG[27570] manager.c: Manager received command 'Command' May 26 08:06:33 DEBUG[27570] manager.c: Manager received command 'Command' May 26 08:06:34 VERBOSE[28367] logger.c: == Connect attempt from '10.34.9.135' unable to authenticate May 26 08:06:46 DEBUG[2855] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found May 26 08:06:59 DEBUG[2855] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' May 26 08:06:59 DEBUG[2855] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Does anybody know if the login name and password used is correct, or what I can try to fix this problem? Werner Werner Terreblanche CONTROL INSTRUMENTS TELEMATICS Tel: +27 21 880 5500 / 5686 (direct) Fax: +27 21 880 1756 Mobile: +27 82 3037669 [EMAIL PROTECTED] www.ci-omnibridge.com Read our disclaimer at http://www.ci-omnibridge.com/contact/disclaimer.aspx ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 301's drop last two digits of dialed number
Hi All, Having a rather annoying problem with the Polycom 301 phones, suspect it to be my dialplan. Basically if you lift the receiver off the handset and dial a number, it will not let you dial a number longer than 10 digits (Can see this being acceptable in US, but in UK its a right pain). As soon as the 10th digit is entered, it starts to dial and the number is invalid. If the phone is left on hook and the number is dialed, it works fine when pressing the 'send' key on the handset as it sends the whole number. Can anyone shed any light on this issue? I thought it could be asterisk is trying to Dial to soon so I added a Wait in the dialplan but it didn't seem to work. Kind regards Jamie Heckford Technical Consultant ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: X100P fails to initialize
>From what I understood Zaptel was ported to the Mac quite some time ago. http://lists.digium.com/pipermail/asterisk-users/2004-October/060872.html Also, TerraSoft sponsored an Asterisk port to YellowDog Linux on PPC - from what I gather, with full Zaptel support. http://www.voip-info.org/wiki/view/Asterisk+Linux+Yellow+Dog I didn't think it's be necessary to run YDL to get Zaptel to work. Does anyone know if this is the case? From what I understand, as long as zaptel compiles it's up to udev and the kernel to do the remaining hardware detection and resource assignment, which should be a distribution-agnostic process. Is it possible that this particular chip used in this particular X100P clone is not supported on LinuxPPC? It is a Motorola 6508 chip, identical to the SM56 winmodem. A picture of the card is here: http://cgi.ebay.ca/Its-Real-X100P-FXO-card-oem-for-Digium-Asterisk-pbx_W0QQitemZ9730539695QQcategoryZ61841QQssPageNameZWDVWQQrdZ1QQcmdZViewItem#ebayphotohostingAppreciative for any help. On 5/25/06, Lachek Butalek [EMAIL PROTECTED] wrote: So I took a chance with an X100P knock-off on eBay. I'm running Asterisk + FreePBX on a PowerMac G3 (beige desktop) using Slackintosh 10.2 and kernel 2.6.16.16. Everything has been fine up until now. I compile the 1.2.5 Zaptel drivers without a problem, get the udev configuration in, modprobe zaptel, and finally modprobe wcfxo. At this point, I get the message:ZT_CHANCONFIG failed on channel 1: No such device or address (6) FATAL: Error running install command for wcfxodmesg gives me:Zapata Telephony Interface Registered on major 196Zaptel Version: 1.2.5 Echo Canceller: MG2Failed to initailize DAA, giving up... wcfxo: probe of :00:0e.0 failed with error -5syslog tells me:May 25 21:28:20 asterisk kernel: Failed to initailize DAA, giving up...May 25 21:28:20 asterisk kernel: wcfxo: probe of :00:0e.0 failed with error -5 ztcfg -vv says:Channel 01: FXS Kewlstart (Default) (Slaves: 01)1 channels configured.ZT_CHANCONFIG failed on channel 1: No such device or address (6)and lspci -vv tells me this:00: 0e.0 Communication controller: Motorola: Unknown device 5608 Subsystem: Motorola: Unknown device Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 24 Region 0: I/O ports at fe000800 [size=256] Region 1: Memory at 81803000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=0mA PME(D0+,D1-,D2+,D3hot+,D3cold-) Status: D0 PME-Enable- DSel=0 DScale=0 PME- One of the reasons I'm running on a PowerMac is specifically because I've been told the X100P cards work well on this type of hardware, since it tends to not have problems with flawed PCI buses and IRQ sharing, but I'm starting to have my doubts. If anyone has experience with X100P cards on PPC, or have any other insights, it would be greatly appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent Callback, how to see wath queue is calling the agent?
Could you just set the variable in the part of the dialplan where they enter the queue and then reference it here?Thanks,KyleOn 5/26/06, Massimo Nuvoli
[EMAIL PROTECTED] wrote:This is the problem:
two QueuesAgent logged in as agentcallback and member of the two queues.When a call come in the queue, asterisk call the extension providedby the agentcallbacklogin.
The need is in the extension to have a variable with the queue id.something like:exten = _6XXX,1,Noop(AGENT ${EXTEN:1})exten = _6XXX,n,UserEvent(FOP_Popup|URL:/cmgr/api/popup?e=${EXTEN:1}cid=${CALLERIDNUM}q=${QUEUE})
exten = _6XXX,n,DIAL(ZAP/g2/1234567${EXTEN:1},30)exten = _6XXX,n,HangupAny idea?Thnks.___--Bandwidth and Colocation provided by
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[Asterisk-Users] my kernel not detect my TDM400P card
Hi all I want to install a TDM400P card. I use fedora core 4 and the version of my kernel is 2.6.11-1.1369_FC4smp. When i type lspci, i have this message: 02:01.0 Network controller: Unknown device e159:0001 how can i fix this problem? Thanks for your help! Serge MESSA OVONO Yahoo! Mail réinvente le mail ! Découvrez le nouveau Yahoo! Mail et son interface révolutionnaire. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail.conf
Is it possible to run a command in the voicemail.conf
file to change the from email-address. This way the user who gets
the email, can reply on the mail just by clicking answer. I want
to do something like this
serveremail='grep
${VM_CIDNUM}) /etc/asterisk/voicemail.conf | cut -d, -f3'
Greetings Jan
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Re: [Asterisk-Users] voicemail.conf
Hi Jan,
maybe externnotify voicemail.conf command may help you to exec an
external script.
Giorgio Incantalupo
Jan Pringels wrote:
Is it possible to run a command in the voicemail.conf file to change
the ‘from’ email-address. This way the user who gets the email, can
reply on the mail just by clicking ‘answer’. I want to do something
like this…
serveremail='grep ${VM_CIDNUM}) /etc/asterisk/voicemail.conf | cut
-d, -f3'
Greetings Jan
// //
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Re: [Asterisk-Users] Limit to number of queues
On 5/26/06, El Flynn [EMAIL PROTECTED] wrote: Hello, Does anyone know the maximum number of queues that can be defined in an Asterisk system? Queues and their members are both stored as linked lists in Asterisk's memory so there really isn't a technical upper limit in the amount you can define. That being said, the algorithm being used, at present, to update the members' device status is not really that efficient and things will get progressively worse as the number of members and queues grow increasing the possibility for a thread deadlock in your system. This is one of a number of reasons why the queueing and agents system is likely to get a real serious overhaul after the 1.4 release. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] misdn problem
I have two HFC ISDN Cards, configured using mISDN on asterisk svn head 1.2 These two cards are connected to 2 ISDN Lines, receiving calls for 50 numbers. Everything is OK on 75 % and bad on 25 % When is bad, In /var/log/asterisk/full I see May 26 09:55:28 WARNING[24410] chan_misdn.c: Extension can never match, so disconnecting May 26 09:55:28 DEBUG[24410] channel.c: Prodding channel 'mISDN/1-1' May 26 09:55:28 DEBUG[24410] channel.c: Scheduling timer at 160 sample intervals May 26 09:55:28 WARNING[24410] chan_misdn.c: Extension can never match, so disconnecting May 26 09:55:28 DEBUG[24410] channel.c: Prodding channel 'mISDN/2-1' May 26 09:55:28 DEBUG[24410] channel.c: Scheduling timer at 160 sample intervals on the asterisk console, if I set misdn set debug 10, I see P[ 1] handle_frm: frm-addr:42000103 frm-prim:3f082 P[ 1] -- lib: NEW_CR Ind with l3id:200ec on this port. P[ 1] -- new_process: New L3Id: 200ec P[ 1] handle_frm: frm-addr:42000103 frm-prim:30582 P[ 1] set_channel: bc-channel:0 channel:1 P[ 1] lib Got Prim: Addr 42000103 prim 30582 dinfo 200ec P[ 1] $$$ find_chan: No channel found for oad:3481303064 dad:0108680550 P[ 0] $$$ find_chan: No channel found with l3id:200ec P[ 1] I IND :SETUP oad:3481303064 dad:0108680550 P[ 1] -- mode:TE cause:16 ocause:16 rad: P[ 1] -- facility:FAC_NONE out_facility:FAC_NONE P[ 1] -- info_dad: onumplan:0 dnumplan:2 rnumplan: P[ 1] -- screen:0 -- pres:0 P[ 1] -- channel:1 caps:Speech pi:0 keypad: P[ 1] -- urate:0 rate:16 mode:0 user1:0 P[ 1] -- pid:336 addr:50010102 l3id:200ec P[ 1] -- b_stid:0 layer_id:50010180 P[ 1] -- bc:81681d4 h:0 sh:0 P[ 1] $$$ find_chan: No channel found for oad:3481303064 dad:0108680550 P[ 1] -- Bearer: Speech P[ 1] -- Codec: Alaw P[ 0] -- * NEW CHANNEL dad:0108680550 oad:3481303064 P[ 1] read_config: Getting Config P[ 1] config_jb: Called P[ 1] -- * CallGrp: PickupGrp: P[ 1] * Queuing chan 0x842e8b0 P[ 1] CONTEXT:from-pstn P[ 1] Tone Indicate: P[ 1] -- Busy P[ 1] misdn_write: * prods us P[ 1] SENDEVENT: stack-nt:0 stack-uperid:4104 P[ 1] I SEND:DISCONNECT oad:3481303064 dad:00108680550 P[ 1] -- mode:TE cause:16 ocause:1 rad: P[ 1] -- facility:FAC_NONE out_facility:FAC_NONE P[ 1] -- info_dad: onumplan:0 dnumplan:2 rnumplan: P[ 1] -- screen:0 -- pres:0 P[ 1] -- channel:1 caps:Speech pi:0 keypad: P[ 1] -- urate:0 rate:16 mode:0 user1:0 P[ 1] -- pid:336 addr:50010102 l3id:200ec P[ 1] -- b_stid:0 layer_id:50010180 P[ 1] -- bc:81681d4 h:0 sh:0 P[ 1] GOT SETUP OK P[ 1] Freeing Msg on prim:30582 P[ 2] handle_frm: frm-addr:42000203 frm-prim:3f082 P[ 2] -- lib: NEW_CR Ind with l3id:40076 on this port. P[ 2] -- new_process: New L3Id: 40076 P[ 2] handle_frm: frm-addr:42000203 frm-prim:30582 P[ 2] set_channel: bc-channel:0 channel:1 P[ 2] lib Got Prim: Addr 42000203 prim 30582 dinfo 40076 P[ 2] $$$ find_chan: No channel found for oad:3481303064 dad:0108680550 P[ 1] $$$ find_chan: No channel found with l3id:40076 P[ 2] I IND :SETUP oad:3481303064 dad:0108680550 P[ 2] -- mode:TE cause:16 ocause:16 rad: P[ 2] -- facility:FAC_NONE out_facility:FAC_NONE P[ 2] -- info_dad: onumplan:0 dnumplan:2 rnumplan: P[ 2] -- screen:0 -- pres:0 P[ 2] -- channel:1 caps:Speech pi:0 keypad: P[ 2] -- urate:0 rate:16 mode:0 user1:0 P[ 2] -- pid:337 addr:50010202 l3id:40076 P[ 2] -- b_stid:0 layer_id:50010280 P[ 2] -- bc:8173dec h:0 sh:0 P[ 2] $$$ find_chan: No channel found for oad:3481303064 dad:0108680550 P[ 2] -- Bearer: Speech P[ 2] -- Codec: Alaw P[ 0] -- * NEW CHANNEL dad:0108680550 oad:3481303064 P[ 2] read_config: Getting Config P[ 2] config_jb: Called P[ 2] -- * CallGrp: PickupGrp: P[ 2] * Queuing chan 0x84a8ea0 P[ 2] CONTEXT:from-pstn P[ 2] Tone Indicate: P[ 2] -- Busy P[ 2] misdn_write: * prods us P[ 2] SENDEVENT: stack-nt:0 stack-uperid:4204 P[ 2] I SEND:DISCONNECT oad:3481303064 dad:00108680550 P[ 2] -- mode:TE cause:16 ocause:1 rad: P[ 2] -- facility:FAC_NONE out_facility:FAC_NONE P[ 2] -- info_dad: onumplan:0 dnumplan:2 rnumplan: P[ 2] -- screen:0 -- pres:0 P[ 2] -- channel:1 caps:Speech pi:0 keypad: P[ 2] -- urate:0 rate:16 mode:0 user1:0 P[ 2] -- pid:337 addr:50010202 l3id:40076 P[ 2] -- b_stid:0 layer_id:50010280 P[ 2] -- bc:8173dec h:0 sh:0 P[ 2] GOT SETUP OK P[ 2] Freeing Msg on prim:30582 P[ 2] MGMT: Short status dinfo 201 P[ 2] MGMT: SSTATUS: L2_ESTABLISH all the (4) channels were idle at call time. Thanks in advance Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 301's drop last two digits of dialed number
Jamie Heckford wrote: Hi All, Having a rather annoying problem with the Polycom 301 phones, suspect it to be my dialplan. This would be incorrect. Basically if you lift the receiver off the handset and dial a number, it will not let you dial a number longer than 10 digits (Can see this being You need to update your digitmap for the Polycom. Look to the sip.cfg in the Polycom tftp directory. Search it for digitmap and update accordingly. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 301's drop last two digits of dialed number
On 26/05/2006, at 7:49 PM, Jamie Heckford wrote: Can anyone shed any light on this issue? I thought it could be asterisk is trying to Dial to soon so I added a Wait in the dialplan but it didn't seem to work. Polycoms have their own dialplan built into the phone. Depending on how you configure your phone (i.e. on the phone, or via the web interface or via FTP), you will have modify the onboard dialplan to allow numbers longer than 10 digits. Hope that helps, Avi -- National Manager - Special Projects Sydney / Melbourne / Canberra / Hobart / London / 2/340 Gore StreetT: +61 (0) 3 9235 5400 Fitzroy, VIC F: +61 (0) 3 9235 5444 3065 W: http://www.squiz.net . Open Source - Own It - Squiz.net .. / ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk.NET authentication problem
Marco youre advise worked like a charm! I put in the IP of my PC and now the authentication works and I can see all events. Thank you very much. J Werner Message: 8 Date: Fri, 26 May 2006 10:35:00 +0100 From: Marco Mouta [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk.NET authentication problem To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 My guess would be to check your manager.conf [admin] secret = amp111 deny=0.0.0.0/0.0.0.0 permit=10.0.0.1/255.255.255.0 read = system,call,log,verbose,command,agent,user write = system,call,log,verbose,command,agent,user The line permit=10.0.0.1/255.255.255.0 should be adjust to your network configurations. Hope it helps! Best regards, Marco Mouta Ps.Please let me know if it worked. Read our disclaimer at http://www.ci-omnibridge.com/contact/disclaimer.aspx ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent Callback, how to see wath queue is calling the agent?
Kyle Sexton wrote:
Could you just set the variable in the part of the dialplan where they
enter
the queue and then reference it here?
Yes, that is the way to do this. Set a variable in the dialplan before
putting the _caller_ into the queue, and prefix the variable with at
least one underscore ('_'). This will cause the variable to be inherited
into the channel that is created to dial the queue member (agent).
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Re: [Asterisk-Users] my kernel not detect my TDM400P card
serge messa wrote: I want to install a TDM400P card. I use fedora core 4 and the version of my kernel is 2.6.11-1.1369_FC4smp. When i type lspci, i have this message: 02:01.0 Network controller: Unknown device e159:0001 how can i fix this problem? There is no problem. The TDM400P does not have a unique PCI vendor/device ID, so it does not show up as a specific device. This does not affect proper operation of the card, only the lspci output. If you run 'update-pciids' you might get a newer database and find an entry other than 'Unknown device'. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent Callback, how to see wath queue is calling the agent?
Kyle Sexton ha scritto: Could you just set the variable in the part of the dialplan where they enter the queue and then reference it here? :-) very simple, tested but not working, and logically i think it is right. In asterisk a variable (dialplan SET) is bound to the incoming channel, but, when the agentcallback is called there is not link with the incoming channel, as the user called must pickup the phone and confirm the call answer. As there is some exception (like CALLERID) i think there is some info missing (or never requested) calling the callback procedure... I think, but my thinking may be wrong. signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent Callback, how to see wath queue is calling the agent?
Kevin P. Fleming ha scritto:
Kyle Sexton wrote:
Could you just set the variable in the part of the dialplan where they
enter
the queue and then reference it here?
Yes, that is the way to do this. Set a variable in the dialplan before
putting the _caller_ into the queue, and prefix the variable with at
least one underscore ('_'). This will cause the variable to be inherited
into the channel that is created to dial the queue member (agent).
Sorry! when i read about the channel variable i forgot to check this
_... :-( i am sorry
Please wait until i made some test, if it is working i post my
configuration here. :-)
Thnks Thnks.
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Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
I looked long and hard at the LAN and it was basically narrowed down to the switches. In this smaller install, several cheapo Dlink ($30) switches de-aggregate a Cisco Catalyst switch. What I noticed was that any phone plugged direcly into the Catalyst did *not* lock up or reboot. Any phone plugged into the crap switches experienced the lockup. So now we are down to the cheap switches themselves. We are nuking the Dlink switches and replacing them with 3com workgroup switches, same as what we use in the large install to good effect, and I fully expect the problem to dissapear. So does anyone have any theories as to what the technical difference between a good switch and a bad or cheapo switch actually is? Lower latency? Better grounding? More cowbell? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] End of migration: adding support for some analog phones
Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN -- Asterisk -- E1 cable -- Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I use a TDM2400P board together with the actual TE410P? Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Remco Barende wrote: On Fri, 26 May 2006, Dave Cotton wrote: On Fri, 2006-05-26 at 10:11 +0200, Remco Barende wrote: Thanks for your input! Previously I was using Nortel 10/100 switches, I replaced them some weeks ago with 3C16479 gbit switches. The phones are connected directly to the gbit switches. By coincidence I dit notice on one phone that in a split second a message appeared 'Ethernet cable disconnected'. Because I have cable unplug set to ignore the conversation was not interrupted and the conversation could continue. But that still doesn't solve the occasional lockup. Looks like you're getting somewhere now. That was my real complaint xyz sucks helps no one. As I said in my reply I've never had such problems with SNOM, perhaps it's because I've always used decent switches. You mean that 3Com switches are not to be regarded as decent switches? At least Snom could have put some remark then that you need a certain brand of switches. If 3Com is not good enough for the phones I would have bought different phones. Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting stuck right at the beginning
I just installed [EMAIL PROTECTED] 2.8 SW on a DELL box. I can connect from my webbrowser to the AMP GUI and can with no problem work with it. The DELL box has 2 NICs and is connected itself to an ADSL router. The firewall is on the external NIC (eth0), the firewall of the router is switched off. In order to connect to the internet, I setup the firewall, packet forwarding, DNS and add routeI can surfe with my laptop on the internet being connected to the internal NIC (eth1). So that should be correct. I am trying now to connect a Grandstream GXP2000 to asterisk but get no connection to the server. I followed Nerd Vittel's procedure step by step, but no way to connect (I am not talking of making calls yet). There is communication between the GXP2000 and asterisk. I set the asterisk server as the NTP (time) server for the phone and the correct date/time is displayed on the phone.Any ideas what I am doing wrong? Thanks a lot in advance for your help ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent Callback, how to see wath queue is calling the agent?
Massimo Nuvoli ha scritto:
Kevin P. Fleming ha scritto:
Kyle Sexton wrote:
Could you just set the variable in the part of the dialplan where they
enter
the queue and then reference it here?
Yes, that is the way to do this. Set a variable in the dialplan before
putting the _caller_ into the queue, and prefix the variable with at
least one underscore ('_'). This will cause the variable to be inherited
into the channel that is created to dial the queue member (agent).
Sorry! when i read about the channel variable i forgot to check this
_... :-( i am sorry
Please wait until i made some test, if it is working i post my
configuration here. :-)
when the channel is received:
exten = 123456789,n,SetVar(__CODA=coda1)
(note: TWO underscore __ and not ONE)
in the callback:
exten = _6XXX,n,UserEvent(FOP_Popup|URL:
/cmgr/api/popup?e=${EXTEN:1}cid=${CALLERIDNUM}q=${CODA})
now is working!
Thnks to everyone helped me!
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[Asterisk-Users] using a billing system
Hello to all, Im trying to use DeadAGI to implement billing with Asterisk2Billing. Before the billing, I had something like: exten = _2,1,Dial(SIP/[EMAIL PROTECTED]) Now, with Asterisk2Billing would be something like this? exten = _2,1,Answer exten = _2,2,Wait,2 exten = _2,3,DeadAGI,a2billing.php exten = _2,4,Wait,2 exten = _2,5,Hangup I tried it and the call is answered bu Asterisk and never dials the destination. :( What do I need to put in the Asterisk configuration in order to make the call and start the billing engine? Thanks Regards Joao Pereira ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
I would like to suggest using any managed switch and hard setting the ports to 100/full I have found that the auto negotiation algorithm is generally to blame on many switches. As an example, connecting a cisco router to a netgear/dlink/3com/etc will geneerate errors on the cisco interface. connecting cisco to cisco does not. connectig netgear/dlink etc does not. But disabling auto makes all play nice together. On May 26, 2006, at 7:38 AM, Rich Adamson wrote: Remco Barende wrote: On Fri, 26 May 2006, Dave Cotton wrote: On Fri, 2006-05-26 at 10:11 +0200, Remco Barende wrote: Thanks for your input! Previously I was using Nortel 10/100 switches, I replaced them some weeks ago with 3C16479 gbit switches. The phones are connected directly to the gbit switches. By coincidence I dit notice on one phone that in a split second a message appeared 'Ethernet cable disconnected'. Because I have cable unplug set to ignore the conversation was not interrupted and the conversation could continue. But that still doesn't solve the occasional lockup. Looks like you're getting somewhere now. That was my real complaint xyz sucks helps no one. As I said in my reply I've never had such problems with SNOM, perhaps it's because I've always used decent switches. You mean that 3Com switches are not to be regarded as decent switches? At least Snom could have put some remark then that you need a certain brand of switches. If 3Com is not good enough for the phones I would have bought different phones. Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. Just curious - what configuration issues did you have in mind? - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hint priority and realtime
Can someone shed some light on why the hint feature was implemented in the priority field that is purely an integer in the rest of the dialplan? There seems to be a conflict with realtime and the hint priority, in order to put in the hints you would have to change the priority column in the database from int to char and give up some performance (since int indexes better and priority is a parameter in the select)? More importantly, can anyone answer these questions; Can the hint priority by put in mysql realtime? Is there truly an impact to changing the priority datatype to char or varchar? If it can not be put in realtime, can the hint priority exist in the same context statically, and the numbered portion of the dialplan in realtime? (making it not so real time) Thanks for any info on this. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Dr. Michael J. Chudobiak wrote: I looked long and hard at the LAN and it was basically narrowed down to the switches. In this smaller install, several cheapo Dlink ($30) switches de-aggregate a Cisco Catalyst switch. What I noticed was that any phone plugged direcly into the Catalyst did *not* lock up or reboot. Any phone plugged into the crap switches experienced the lockup. So now we are down to the cheap switches themselves. We are nuking the Dlink switches and replacing them with 3com workgroup switches, same as what we use in the large install to good effect, and I fully expect the problem to dissapear. So does anyone have any theories as to what the technical difference between a good switch and a bad or cheapo switch actually is? Lower latency? Better grounding? More cowbell? By far, the majority of switches on the market today will work just fine for VoIP. Past professional experience dictates (in my mind) that a managed switch is the only reasonable approach for any network larger then a home office. There are some inexpensive switches being sold that are less then adequate for business use. For example, Dell rebranded and sold some switches about two years ago that would reboot if an html packet hit the manager IP address; didn't even have to be a crafted packet. Cabletron sold a number of models that would auto reboot at random intervals. HP had some issues with early firmware that essentially resulted in reboots (it was fixed in later firmware versions). Our company conducts professional network performance, security, and voip readniness assessments, and have worked with corporations and institutions in over 40 US states in the last 12 years. We constantly see folks making assumptions about how switches function that are far less then accurate. One example is leaving switch ports to auto negotiate duplex settings. Roughly 50% of the time the switch (and/or device attached to the switch) will get it wrong; one will be full duplex while the other ends up half duplex. That one item will have a serious impact on voip quality. The only way to ensure a solid network infrastructure is to use switches that are manageable, and there are now lots of inexpensive switches on the market that are manageable. In very general terms, the higher the cost of the switch, the more functionality one receives. Also in very general terms, the larger the network, the more functionality one needs within the switches. In other words, a network with several hundred switch ports likely requires switches with the capability of supporting vlans, packet queuing/prioritization, etc. Small networks (eg, low traffic volumes) in most cases do not need those same functions. So, your choice of switches is highly dependent on the size of network that your working with. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Dr. Michael J. Chudobiak wrote: Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. Just curious - what configuration issues did you have in mind? - Mike Replacing it with a Catalyst? Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Need a recomendations and config samples. FXS-SIP terminal with 4 ports.
Hi, folks. I want to buy FXS-SIP terminal with 4 ports (up to 250$). Do you have any recomendations and Asterisk configurations samples for such devices. Any pitfalls? Actually i realy don't know what to buy? -- = = Best regards, Nikolay Pavlov. = = ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP provider for Turkey from India with Asterisk
Hi Friends, At present, I am using VoIPJET.COM provider for make calls to USA. I have two doubts. 1) I am unable to make call to UK Mobile phone. Why? 2) I want to make calls to "Turkey" country from "India". With VoIPJET, I am unable to make call to "Turkey" and unable to find VoIP provider for Turkey. Please tell me VoIP Provider for Turkey from India. Looking forward for your response. ThanksRegards, Chandramouli Sneak preview the all-new Yahoo.com. It's not radically different. Just radically better. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP provider for Turkey from India with Asterisk
Plainvoip has a very good A-Z and I have found they are fairly inexpensive. They also offer TollFree orig and some local dids. www.plainvoip.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Crazy Boy Sent: Friday, May 26, 2006 9:21 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoIP provider for Turkey from India with Asterisk Hi Friends, At present, I am using VoIPJET.COM provider for make calls to USA. I have two doubts. 1) I am unable to make call to UK Mobile phone. Why? 2) I want to make calls to Turkey country from India. With VoIPJET, I am unable to make call to Turkey and unable to find VoIP provider for Turkey. Please tell me VoIP Provider for Turkey from India. Looking forward for your response. ThanksRegards, Chandramouli Sneak preview the all-new Yahoo.com. It's not radically different. Just radically better. This email was scanned by: Mcafee GroupShield CONFIDENTIAL DISCLAMER All information provided in this email is considered confidential and proprietary of Planet Telecom, Inc. and Telecenter Inc. Use of this information by anyone other than the recipient or sender will be considered in breach of agreement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Dr. Michael J. Chudobiak wrote: Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. Just curious - what configuration issues did you have in mind? A partial list of issues that we've seen in the last 12 years include: - auto negotiation of duplex settings (mismatch) - spanning tree disabling ports for first 30 seconds after any link state change (some attached devices don't like that) - spanning tree loops that end up isolating devices from the backbone (spanning tree is usually implemented by the manufacture by default) - various switch manufacturers have licensed/implemented cisco's discovery protocol, and the user doesn't realize some equipment attached to such ports actually use the cdp data to change port configuration, while other devices might barf on those packets. - assumptions that all switches operate at wire speeds and buffer packets (eg, no such thing as a switch buffer; packets will be dropped under high load conditions) - distributing vlans across multiple switches where assumptions are made relative to what happens when two or more vlans are transporting traffic volumes that when combined exceed a trunk's port speed (eg, don't forget about broadcast storms). - switch forwarding tables that are too small (eg, workgroup switches) and the table fills, essentially turning the switch into a hub - bad assumptions relative to rate limiting broadcast and multicast packets, and how that impacts normal traffic. - etc, etc. In the case of switch forwarding tables, its very common to see experienced engineers (and others) assume a workgroup switch can be used in a large network environment where 23 ports are used for devices within a small workgroup. However, all switches on the market listen for traffic from any source (including upstream link), and populates the switch forwarding table with the mac addresses observed. Most workgroup switches are limited to 1,024 table entries (sometimes less), and when that table is full, does something that is vendor dependent. Some vendors actually clear the table (resulting in the switch operating as a hub until the table is rebuilt again), while other vendors replace the oldest entries with the newest mac address observed. Some vendors will timeout table entries in very short periods of time. The end result from those actions is packets appearing on switch ports for which the attached device has no need to hear (eg, increases the packet traffic on a per port basis). There are lots of other cases where a switch will forward multicast packets to all ports (eg, poor/incorrect multicast support), and the device attached to the port isn't designed to handle such packets. For example, MS systems (and others) spew UPNP multicast packets looking for or advertising gateways and other network resources. If a Snom device hasn't been programmed correctly to ignore such packets, it might roll over (I don't have a clue if that example is reasonable or even accurate; its just an example only). Changing from one vendor's switch to another might lead one to believe the switch was at fault, when in fact the problem is more related to the switch implementor not properly configuring the first switch. (And, in most cases, the implementor doesn't have a clue what type of packets are flowing across the network, let along which ones result in problems for attached devices.) R. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Andrew D Kirch wrote: Dr. Michael J. Chudobiak wrote: Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. Just curious - what configuration issues did you have in mind? - Mike Replacing it with a Catalyst? Most of the catalyst switches are pretty good. Some of the older ones have had problems with truly supporting traffic volumes that approach 100% of a port's speed. Some catalyst switches do have queue/prioritization. The less expensive ones only support three queues while more expensive ones support greater numbers of queues. Some support the bits in the IP packet header that were intended to influence priority, while other models ignore those bits but implement prioritization on a per-port basis (which basically assumes one device per port). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
More cowbell? Shit, you owe me a new keyboard! Funniest thing I've *ever* read on the list. I've experienced the auto-negotiate issue with Snom's before. I forgot to mention that we make it part of our standard install to force 100baseT-full. I've also noticed the Catalyst does the spanning-tree thing and waits up to 30 seconds before enabling the port - this can cause problems with Snoms because they boot before the Catalyst enables the port, causing registration to fail. Then you warm-boot the Snom and everything's OK. One last interesting tidbit: We have a *lot* of Dell Dimensions with super craptastic embedded Ethernet. They will auto negotiate with a Snom (plugged into the PC port) to 100baseT full, but then you can't ping or TX past the phone itself. Oddly enough, it gets an IP from our DHCP server OK. Forcing the Dell to 100baseT full, half, or even 10 full works 100% of the time. This never happens on any kind of decent Ethernet card like an 82557 chip or 3com. If we have an Optiplex, it *just works* ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCI Problems
Sean Cook wrote: Rob Lith wrote: Does the sangoma handle sharing interrupts in some other way? from: http://www.voip-info.org/wiki/view/Sangoma There are no known compatibility issues (IRQ, IO etc) with ANY Sangoma hardware and ANY make/brand of PC/server- NONE The pci bus was designed to share interrupts between multiple devices. Sangoma cards generally play very nice in that environment while some of the Digium cards do not like shared interrupts at all. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
On Fri, 26 May 2006, Rich Adamson wrote: You mean that 3Com switches are not to be regarded as decent switches? At least Snom could have put some remark then that you need a certain brand of switches. If 3Com is not good enough for the phones I would have bought different phones. Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. The 3C16479 is a non-configurable, non-managed 3Com workgroup gbit switch. It is directly connected to the asterisk server with one cable, the phones are connected to the other ports. There is nothing to configure on the switch. Maybe I need to change my opinion, it's not only the firmware that sucks, if the ethernet chip on the phone is this oversensitive I guess the same would apply for the hardware. There is just no valid reason why the phone would need to lockup or reboot even if the network connection would be problematic, no matter what. That is just poor design, not a feature. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
>From my point of view, using cheap or expensive switch is not the point.The point is what kind of switch implementation Snom phones require ?.Up to now, it seems that problems relate to auto-negociation. Would it be possible for anyone to check that, comparing fixed and auto-negociated behaviours on the same cheap or descent switch ?Cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] End of migration: adding support for some analog phones
2006/5/26, Mimmus [EMAIL PROTECTED]: Hi,during gradual migration to Asterisk, I put Asterisk in front of a legacyAlcatel PBX:PRI PSTN -- Asterisk -- E1 cable -- Alcatel PBXAfter successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some olderstuff. What's the best choice? A channel bank or a TDM2400P card?Can I use a TDM2400P board together with the actual TE410P?Thanks --Domenico ViggianiFrom many inputs, channel bank seems to be the more reliable solution today as you cannot get bridging inside TDM2400 yet.I've been told this bridging feature is planned but not commited. PS: How many users were at start connected to Alcatel PBX ? What did you do for voicemail during migration ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail.conf
No you can't. We've actually got a patch that does exactly that. It uses
realtime heavily, mainly to pull the information directly from the
database. If you're interested, let me know :)
On Fri, 26 May 2006, Jan Pringels wrote:
Is it possible to run a command in the voicemail.conf file to change the
'from' email-address. This way the user who gets the email, can reply on the
mail just by clicking 'answer'. I want to do something like this.
serveremail='grep ${VM_CIDNUM}) /etc/asterisk/voicemail.conf | cut -d,
-f3'
Greetings Jan
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
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Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Remco Barende wrote: On Fri, 26 May 2006, Rich Adamson wrote: You mean that 3Com switches are not to be regarded as decent switches? At least Snom could have put some remark then that you need a certain brand of switches. If 3Com is not good enough for the phones I would have bought different phones. Blaming the 3com switch is very likely to be the wrong root cause. High probability the 3com was not configured properly for the phone. The 3C16479 is a non-configurable, non-managed 3Com workgroup gbit switch. It is directly connected to the asterisk server with one cable, the phones are connected to the other ports. There is nothing to configure on the switch. The switch is doing auto negotiation, whether you can see it or not. That's exactly why I'd never use an unmanaged switch for anything that is critical. Gig in this case has no value whatsoever. Maybe I need to change my opinion, it's not only the firmware that sucks, if the ethernet chip on the phone is this oversensitive I guess the same would apply for the hardware. Part of the problem with this half vs full duplex is there are no commonly implemented industry standards for negotiating a correct setting. Essentially, the switch port and the attached device auto negotiates at the same time, and one device sees what it thinks is half duplex when the other device is in the middle of its negotiation process. In most cases, statically defining one of the two is sufficient, but to be 100% accurate from a performance perspective, both should be statically defined. Gig ports that truly operate at gig speeds is not an issue as there is no such thing as half duplex gig. But, if the attached devices only operate at 10/100 speeds, the switch still has to negotiate the half vs full duplex. There is just no valid reason why the phone would need to lockup or reboot even if the network connection would be problematic, no matter what. That is just poor design, not a feature. I'd agree with that 1000%. I stopped using snom products with the 200 for that very reason (eg, lack of testing and quality control). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hints/subscriptions accross IAX
(I hope this isn't html - Thunderbird is so annoying) I'm new to using hints/subscriptions on * so please be patient with me. I have two * systems in different geographic locations, connected via IAX Location1 has a Polycom 600 and a GXP-2000 phone Location 2 has a single GXP-2000. With the latest GS firmware, at Location1 I've managed to get an LED to light up on the GS phone when a line on the Polycom is in use. This is great. But I need to get an LED to light up on a GS in Location2 when a line on the Polycom at Location1 is in use. Is this possible? If so, can anybody give me any pointers as to how? Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
Colin Anderson wrote: More cowbell? Shit, you owe me a new keyboard! Funniest thing I've *ever* read on the list. I've experienced the auto-negotiate issue with Snom's before. I forgot to mention that we make it part of our standard install to force 100baseT-full. I've also noticed the Catalyst does the spanning-tree thing and waits up to 30 seconds before enabling the port - this can cause problems with Snoms because they boot before the Catalyst enables the port, causing registration to fail. Then you warm-boot the Snom and everything's OK. The same spanning tree issue (not forwarding packets for 30 to 60 seconds) is also a problem with most of the newer PC systems (particularly with MS O/S) as the system boots up quicker then when the switch is ready to forward traffic. An MS O/S system begins broadcasting for domain controllers (etc) before the switch is ready to forward traffic resulting in some very strange problems that most Sys Admins diagnose incorrectly. One last interesting tidbit: We have a *lot* of Dell Dimensions with super craptastic embedded Ethernet. They will auto negotiate with a Snom (plugged into the PC port) to 100baseT full, but then you can't ping or TX past the phone itself. Oddly enough, it gets an IP from our DHCP server OK. Forcing the Dell to 100baseT full, half, or even 10 full works 100% of the time. This never happens on any kind of decent Ethernet card like an 82557 chip or 3com. If we have an Optiplex, it *just works* Right on! But, its not just the Dell products. There are a fair number of other products with the same issue, and a few drivers that have half/duplex backwards (set it to half and the interface operates in full, or, setting to either half or full fails but auto works in full duplex just fine). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCI Problems
On Thursday 25 May 2006 16:11, Sean Cook wrote: What could be the other causes? I have exhausted everything I know how to do. PCI sharing explains it (whether or not it is infact the problem). This card shares the BIOS assigned interrupt with the network card... Audio problems can come for a variety of reasons. They are caused by (but not limited to) things such as - IRQ sharing with another device with a shitty driver or poor hardware - Poor/inconsistent PCI bus behaviour and timing - overloaded CPU or poor kernel parameters which cause timing problems - shitty hardware or drivers which can lock out IRQs for a long time - buggy drivers for the TDM or ethernet hardware - bad PCI tuning with setpci or kernel parameters, latency timers especially - other hardware (PCI bus controller, north or south bridge) issues - faulty hardware - poor cabling (either TDM side or ethernet side) IRQ sharing is often blamed for audio problems but the fact of the matter is that IRQ sharing is *NOT* an issue if the hardware that is sharing the IRQ (and the drivers for that hardware) plays nicely and reacts to the IRQ quickly. PCI is DESIGNED to share IRQs. The trouble comes when vendors take old ISA hardware, port it to PCI and/or don't ensure that they not only share IRQs properly but also do not ensure that their drivers check that their hardware caused the IRQ and react to IRQs quickly. There is NOTHING inherently wrong with sharing IRQs. The IRQ handler needs to check the hardware to see if it was their hardware that generated the IRQ and get the hell out if not. A lot of (poor) drivers do NOT do this. The driver either assumes that the IRQ MUST have been generated by the hardware (which can cause a host of weird problems), or the check takes so long that it causes trouble for the card that DID generate the IRQ. Digium's hardware is more sensitive to IRQ sharing trouble than other hardware for two very simple reasons. The first is that the TDM cards have no real buffering. If the data is not taken from the register it will quickly be overwritten by the next block of data. This is analogous to the old 16450 UARTs of yore. They had a receiver shift register and a 1-byte receiver buffer. If you didn't get the data out of the buffer before the next byte had shifted in, the new byte would be transferred to the buffer and you'd get an overrun error. The 16550 replaced the 1-byte receive buffer with a 16-byte FIFO (IIRC) -- you could trigger an IRQ after the FIFO had filled 'x' bytes, and then service the IRQ, retrieving all bytes received in one fell swoop. And if your IRQ service routine got a little delayed it was no big deal because there was room for another byte or two before you started losing data. This allowed the IRQ volume on busy serial applications to be far lower (up to 16x lower) than before, which allowed for better system utilization. Digium's hardware is like the old 16450. There is no FIFO. This was done consciously, and is not necessarily a bad design -- TDM is VERY sensitive to latencies. The more delay you have, the worse things like echo become. Bringing TDM data into the PC is already pretty laggy. Adding more delay with FIFOs isn't necessarily a good thing. (I would argue that having a 16 byte FIFO and triggering the IRQ on the first position would not be a bad thing nor would it introduce any latency, but that's me. I'd change a few things about Digium's hardware, but there is no arguing at their success.) So back to the problem at hand: if there is significant delay between the IRQ and the IRQ service, you lose data. This leads to chirping/clicking and in the case of T1, HDLC/framing errors, dropped links and bouncing D channels (for PRI). The second reason is that Digium's drivers do a LOT of work in the IRQ handler. Essentially they are poor PCI neighbours. In the past (I have not checked this recently) all of the echo cancellation and heavy lifting was done right inside the IRQ handler, with interrupts disabled. This caused their IRQ service time to be lengthy, and until interrupts are enabled again you essentially lock out any other driver from servicing its hardware. (Basically Digium's drivers do to other drivers what Digium's drivers can't stand to have done to it.) Contrast this with Sangoma's drivers, which get the data into system RAM, set a flag (softIRQ?) and then get the hell out of the IRQ context as quickly as possible. Then whenever the CPU gets time to do it, the driver takes the data and processes it OUTSIDE of the IRQ context. Whether this is better or worse for performance is under debate, but there is absolutely no question that doing it this way makes their products better PCI neighbours. This is a rather lengthy post, and I am sure that others will post contradictory or corrective responses, which I welcome. The jist of the post, however, is that there are far more things that can
[Asterisk-Users] Two questions about [EMAIL PROTECTED] and backups.
First question, is there a forum for [EMAIL PROTECTED] specific questions? I've asked what must have been questions about [EMAIL PROTECTED] here and gotten some indication they weren't welcome. Second, does anyone know what files need to be backed up? I don't need to back up the entire system since I can reinstall from the CD in fairly quick order, however, other than the files in /etc, where else does asterisk keep files that need to be backed up? Thanks, Jim. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: End of migration: adding support for some analogphones
This is whatam doing for voicemail during my
transition.
My pbx send the 9 out on all calls. (made my
asterisk configs easier.)
All of my extensions start with 5. asterisk
extension are 56XX and 57XX, Legacy extensions are 51XX and 52XX.
I added the below lines to my
dialplan.
exten = _92XXX,1,AGI(calleridname.agi)exten
= _92XXX,2,Macro(vm,5${EXTEN:2})
Then I set the call forward busy and no-ans, to the
legacy phone's extension less the 5 pretended by a 2.
So the call forward for extension 5122 is
2122.
The above dailplan sends 2122 to the voicemail box
of 5122.
This was the simplest solution I could
find.
I am using FreePBX for configs, so I had to make a
custom extension with voicemail set to dial zap/g2/5122 to be able to keep the
configs in FreePBX.
If not using FreePBX, I think you would only have
to add the Legacy extension in voicemail.conf.
-- -- Steven
http://www.glimasoutheast.org
"Olivier Krief" [EMAIL PROTECTED] wrote in
message news:[EMAIL PROTECTED]...2006/5/26,
Mimmus [EMAIL PROTECTED]:
Hi,during
gradual migration to Asterisk, I put Asterisk in front of a
legacyAlcatel PBX:PRI PSTN -- Asterisk
-- E1 cable -- Alcatel PBXAfter successful
deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to
keep some of analog lines to support modem, fax and some olderstuff.
What's the best choice? A channel bank or a TDM2400P card?Can I use a
TDM2400P board together with the actual TE410P?Thanks
--Domenico Viggiani
From many inputs, channel bank seems to be the more reliable solution
today as you cannot get bridging inside TDM2400 yet.I've been told this
bridging feature is planned but not commited. PS: How many users were
at start connected to Alcatel PBX ? What did you do for voicemail during
migration ?
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Re: [Asterisk-Users] USB headsets?
On Thursday 25 May 2006 17:48, mustardman29 wrote: Just remember that USB audio devices such as a USB headset increases CPU usage compared to standard audio. It's probably not much of a problem for modern processors but I don't have any direct experience with them to confirm this. Not necessarily. USB audio devices can take audio and process it on their own CPU instead of requiring it in raw formats. Some PCI audio cards can do this too though. Basically the difference should be negligible. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] large duration calls
Hello mates, im having calls of about 120 o 130 minutes in my accounting DB but these are calls not made by users.I guess my asterisk is not catching some BYE requests and after some timeout it hangs up the call.Is this issue known? Is there a way to trace this problem? Im using Asterisk 1.0.10 on a i686 linux and mysql for the accounting. I have this problem with different protocols, like SIP and Zap.If you need another specs or something, askme.Cheers, Francisco. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Two questions about [EMAIL PROTECTED] and backups.
[EMAIL PROTECTED] is welcome as long as you are referring to the asterisk portion of it and not the GUI or dialplans that make [EMAIL PROTECTED] different from the typical asterisk. I believe [EMAIL PROTECTED] offers a backup button that will backup all pertinent files for you. I.e. dialplan, modules, sip.conf, etc. bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Lynch Sent: Friday, May 26, 2006 10:39 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Two questions about [EMAIL PROTECTED] and backups. First question, is there a forum for [EMAIL PROTECTED] specific questions? I've asked what must have been questions about [EMAIL PROTECTED] here and gotten some indication they weren't welcome. Second, does anyone know what files need to be backed up? I don't need to back up the entire system since I can reinstall from the CD in fairly quick order, however, other than the files in /etc, where else does asterisk keep files that need to be backed up? Thanks, Jim. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1443 (20060314) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Two questions about [EMAIL PROTECTED] and backups.
Jim,There are SourceForge.net forums for [EMAIL PROTECTED] where you'll probably find better answers to your AAH questions. They are located here: https://sourceforge.net/forum/?group_id=123387 In terms of backup, AAH has a built-in backup feature as part of the FreePBX GUI. Set your schedule, tell it which components (configs, voicemails, etc) you want to backup, and it'll generate a tarball that you can pull off the system at your leisure. AlexOn 5/26/06, Jim Lynch [EMAIL PROTECTED] wrote: First question, is there a forum for [EMAIL PROTECTED] specific questions?I've asked what must have been questions about [EMAIL PROTECTED] here andgotten some indication they weren't welcome.Second, does anyone know what files need to be backed up? I don't need to back up the entire system since I can reinstall from the CD in fairlyquick order, however, other than the files in /etc, where else doesasterisk keep files that need to be backed up?Thanks,Jim. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] large duration calls
I also had this same problem with an older version of asterisk. The issue disappeared when I upgraded. Upgrade to a newer version see if it still exists. bp From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Francisco Seratti Sent: Friday, May 26, 2006 10:51 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] large duration calls Hello mates, im having calls of about 120 o 130 minutes in my accounting DB but these are calls not made by users. I guess my asterisk is not catching some BYE requests and after some timeout it hangs up the call. Is this issue known? Is there a way to trace this problem? Im using Asterisk 1.0.10 on a i686 linux and mysql for the accounting. I have this problem with different protocols, like SIP and Zap. If you need another specs or something, askme. Cheers, Francisco. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCI Problems
Andrew Kohlsmith wrote: On Thursday 25 May 2006 16:11, Sean Cook wrote: What could be the other causes? I have exhausted everything I know how to do. PCI sharing explains it (whether or not it is infact the problem). This card shares the BIOS assigned interrupt with the network card... Audio problems can come for a variety of reasons. They are caused by (but not limited to) things such as - IRQ sharing with another device with a shitty driver or poor hardware - Poor/inconsistent PCI bus behaviour and timing - overloaded CPU or poor kernel parameters which cause timing problems - shitty hardware or drivers which can lock out IRQs for a long time - buggy drivers for the TDM or ethernet hardware - bad PCI tuning with setpci or kernel parameters, latency timers especially - other hardware (PCI bus controller, north or south bridge) issues - faulty hardware - poor cabling (either TDM side or ethernet side) IRQ sharing is often blamed for audio problems but the fact of the matter is that IRQ sharing is *NOT* an issue if the hardware that is sharing the IRQ (and the drivers for that hardware) plays nicely and reacts to the IRQ quickly. PCI is DESIGNED to share IRQs. The trouble comes when vendors take old ISA hardware, port it to PCI and/or don't ensure that they not only share IRQs properly but also do not ensure that their drivers check that their hardware caused the IRQ and react to IRQs quickly. There is NOTHING inherently wrong with sharing IRQs. The IRQ handler needs to check the hardware to see if it was their hardware that generated the IRQ and get the hell out if not. A lot of (poor) drivers do NOT do this. The driver either assumes that the IRQ MUST have been generated by the hardware (which can cause a host of weird problems), or the check takes so long that it causes trouble for the card that DID generate the IRQ. Digium's hardware is more sensitive to IRQ sharing trouble than other hardware for two very simple reasons. The first is that the TDM cards have no real buffering. If the data is not taken from the register it will quickly be overwritten by the next block of data. This is analogous to the old 16450 UARTs of yore. They had a receiver shift register and a 1-byte receiver buffer. If you didn't get the data out of the buffer before the next byte had shifted in, the new byte would be transferred to the buffer and you'd get an overrun error. The 16550 replaced the 1-byte receive buffer with a 16-byte FIFO (IIRC) -- you could trigger an IRQ after the FIFO had filled 'x' bytes, and then service the IRQ, retrieving all bytes received in one fell swoop. And if your IRQ service routine got a little delayed it was no big deal because there was room for another byte or two before you started losing data. This allowed the IRQ volume on busy serial applications to be far lower (up to 16x lower) than before, which allowed for better system utilization. Digium's hardware is like the old 16450. There is no FIFO. This was done consciously, and is not necessarily a bad design -- TDM is VERY sensitive to latencies. The more delay you have, the worse things like echo become. Bringing TDM data into the PC is already pretty laggy. Adding more delay with FIFOs isn't necessarily a good thing. (I would argue that having a 16 byte FIFO and triggering the IRQ on the first position would not be a bad thing nor would it introduce any latency, but that's me. I'd change a few things about Digium's hardware, but there is no arguing at their success.) So back to the problem at hand: if there is significant delay between the IRQ and the IRQ service, you lose data. This leads to chirping/clicking and in the case of T1, HDLC/framing errors, dropped links and bouncing D channels (for PRI). The second reason is that Digium's drivers do a LOT of work in the IRQ handler. Essentially they are poor PCI neighbours. In the past (I have not checked this recently) all of the echo cancellation and heavy lifting was done right inside the IRQ handler, with interrupts disabled. This caused their IRQ service time to be lengthy, and until interrupts are enabled again you essentially lock out any other driver from servicing its hardware. (Basically Digium's drivers do to other drivers what Digium's drivers can't stand to have done to it.) Contrast this with Sangoma's drivers, which get the data into system RAM, set a flag (softIRQ?) and then get the hell out of the IRQ context as quickly as possible. Then whenever the CPU gets time to do it, the driver takes the data and processes it OUTSIDE of the IRQ context. Whether this is better or worse for performance is under debate, but there is absolutely no question that doing it this way makes their products better PCI neighbours. This is a rather lengthy post, and I am sure that others will post contradictory or corrective responses, which I welcome. The jist of the post, however, is that there
Re: [Asterisk-Users] Two questions about [EMAIL PROTECTED] and backups.
You might try these sites: http://sourceforge.net/forum/forum.php?forum_id=420324 Backup has been discussed many times here. Unfortunately, the SF forums suck in terms of searching. http://www.freepbx.org/ http://aussievoip.com.au/wiki/index.php?page=FreePBX https://sourceforge.net/projects/amportal/ http://voipspeak.net/forum/ http://nerdvittles.com/ and sometimes, http://forums.whirlpool.net.au/forum-threads.cfm?f=107 Doug At 09:38 AM 5/26/2006, you wrote: First question, is there a forum for [EMAIL PROTECTED] specific questions? I've asked what must have been questions about [EMAIL PROTECTED] here and gotten some indication they weren't welcome. Second, does anyone know what files need to be backed up? I don't need to back up the entire system since I can reinstall from the CD in fairly quick order, however, other than the files in /etc, where else does asterisk keep files that need to be backed up? Thanks, Jim. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] End of migration: adding support for some analog phones
I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I use a TDM2400P board together with the actual TE410P? As far as I know, Digium doesn't support FAX through TDM2400P, even less a modem call. I had to take one analog line out of my huntgroup and reserve it for modem calls since I could not make it work using the TDM2400P. Note that I was going from an FXS to an FXO. I've read that using a PRI and a good channel bank should work. hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE406P - MFC/R2
Fernando Lujan wrote: Steve Underwood wrote: The problem is almost solved. The card was configured as a T1 interface, the selled came and jumped it. Now I have the following problem. When I call from my legacy pbx, appear a event: *CLI May 26 12:04:09 WARNING[5215]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/31 - 0001 [1/ 1/Idle /Idle ] May 26 12:04:09 WARNING[5215]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/31 Detected May 26 12:04:09 WARNING[5215]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/31 Making a new call with CRN 32769 May 26 12:04:09 WARNING[5215]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/31 1101 - [2/ 2/Idle /Idle ] May 26 12:04:09 WARNING[5215]: chan_unicall.c:2644 handle_uc_event: Unicall/31 event Detected May 26 12:04:54 WARNING[5215]: chan_unicall.c:627 unicall_report: MFC/R2 UniCall/31 - 1001 [2/ 2/Seize ack /Seize ack] But this event, doesn't enter in the context which I configure in the unicall.conf file. Do I need to change something else? Thanks again. Fernando Lujan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 + port translation
Hi all, I'm having trouble with incoming IAX2 calls on 1.2.7.1 - mostly they work, but sometimes the caller just gets dead air or disconnects. IAX2 debugs show HANGUP and INVALID codes in these cases, rather than a proper RINGING transaction. My firewall is doing NAT, and changing the source port from 4569 to something else - my IAX2 provider suggested this might be a problem. Is it? Should this work: steerpike*CLI iax2 show registry Host UsernamePerceived Refresh State 64.26.157.230:45698886708729 64.26.155.62:14353 60 Reg 64.26.157.230:45696134827945 64.26.155.62:14353 60 Reg 64.26.157.230:45696136866597 64.26.155.62:14353 60 Reg 64.26.157.230:45696136866675 64.26.155.62:14353 60 Reg There are four DIDs, and all are registered to an odd port (14353). Is this OK? (I am using a Sonicwall TZ170 with Enable Consistent NAT on). - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: American Telecom Approved by FCC to Certify DECT Phones in US
http://www.wirelessiq.com/content/newsfeed/7319.html Im surprised, I thought DECT was already available in the USA from my days selling this at Ericsson Australia back in 1995. Can someone confirm that they arent already available? Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX2 + port translation
Dr. Michael J. Chudobiak wrote: Hi all, I'm having trouble with incoming IAX2 calls on 1.2.7.1 - mostly they work, but sometimes the caller just gets dead air or disconnects. IAX2 debugs show HANGUP and INVALID codes in these cases, rather than a proper RINGING transaction. My firewall is doing NAT, and changing the source port from 4569 to something else - my IAX2 provider suggested this might be a problem. Is it? Should this work: steerpike*CLI iax2 show registry Host UsernamePerceived Refresh State 64.26.157.230:45698886708729 64.26.155.62:14353 60 Reg 64.26.157.230:45696134827945 64.26.155.62:14353 60 Reg 64.26.157.230:45696136866597 64.26.155.62:14353 60 Reg 64.26.157.230:45696136866675 64.26.155.62:14353 60 Reg There are four DIDs, and all are registered to an odd port (14353). Is this OK? (I am using a Sonicwall TZ170 with Enable Consistent NAT on). - Mike If memory serves me properly what you are showing looks correct. You server is registering to your provider on port 4569 as it should. Their server is seeing you register from 64.26.155.62 and using the prt 14353 which is the port that your firewall has given that outgoing connection. Possibly that the firewall is removing that connection port after some time and your provider cannot get back to your box? Try setting the reregistration time lower than 60 and see if it helps. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: End of migration: adding support for someanalogphones
Olivier Krief [EMAIL PROTECTED] wrote: PS: How many users were at start connected to Alcatel PBX ? What did you do for voicemail during migration? I had ~110 extensions. During migration, I simply avoid to give Asterisk goodies to Alcatel users. Every extension migrated to VoIP could be reached from remaining analog users by a 'follow-me' on the Alcatel: ext -- external line direct engagement codeext i.e on my system: #30ext. In any case, giving voicemail to Alcatel users is definitely possible and Steven's suggestion is an example. Bye Domenico ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] End of migration: adding support for some an alog phones
Nuthin beats an Atlas: http://www.adtran.com/adtranpx/Doc/0/TUA2HMOPDK3KN6S9LM1FH91169/61200305L2-8 A.pdf Telephony Swiss army knife. You can make it do anything. Be prepared to crap your pants when you see the price, though. -Original Message- From: Time Bandit [mailto:[EMAIL PROTECTED] Sent: Friday, May 26, 2006 9:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] End of migration: adding support for some analog phones I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I use a TDM2400P board together with the actual TE410P? As far as I know, Digium doesn't support FAX through TDM2400P, even less a modem call. I had to take one analog line out of my huntgroup and reserve it for modem calls since I could not make it work using the TDM2400P. Note that I was going from an FXS to an FXO. I've read that using a PRI and a good channel bank should work. hth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AstriCon
We attended the Astricon in California, US last year. Although it was not what we expected, we did feel like we gained enough knowledge to make it worth the time and expense to attend. Good luck and let us know how you like the show if you end up attending. --Todd -- VoIP Street DID origination services with support you can count on! http://www.VoIPstreet.com - Original Message - From: Mimmus [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, May 26, 2006 9:06 AM Subject: [Asterisk-Users] AstriCon Hi, I live in Italy and I'm planning to go to the next AstriCon conference in London. Can someone on this list provide me with some detail of previous exhibition? I'd like to have some idea of what I'm running into... Thanks -- Domenico Viggiani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCI Problems
On Friday 26 May 2006 11:15, Rich Adamson wrote: Have you dug into the TDM400 far enough to know whether the common complaints are associated with a hardware design issue, TigerJet issue, or driver? (eg, can any of the issues truly be addressed?) My personal opinion is that the TJ320 (the PCI interface chip) is crap. Total, utter, complete, absolute crap. It can work, yes, but it's first and foremost an economy PCI interface. It's like the Plexor PCI interfaces that let you migrate your ISA design to PCI by plugging a PLX9052 between your ISA interface and the PCI interface. It's a cheap way to get your design on to PCI, but it is very inflexible and VERY ... yuck. My THEORY is that Mark started using it because it was easy to use, but when he went to full-scale production and started selling these things he very quickly discovered how different PCI bridges can be and how much trouble the TJ320 can cause for a rapidly-growing hardware company. The Xilinx Spartan II is much better and far more flexible (but much more expensive, too). I don't doubt he has either personally or had is hardware team research alternatives. They're a very busy and very, VERY smart bunch of people. Mark has brought PCI hardware to the public and supports it, which is something I have not done in any kind of volume. I'd love to hear a detailed, technical response about what they've discovered is wrong, what they are going to do, and where their new products are headed, but I'd also like a 50 acre farm and some horses... :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK experts only. BT Outgoing caller ID not showing
Hi Just moved offices in the UK and moved our Asterisk box from old one to new one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces. At both offices we had one standard number and a DDI range, routed with Asterisk. We'd set up the configuration so each idefisk set its own caller ID which then got sent by the extensions.conf script. Worked fine at old place but in new place the only number which is received is the central switchboard number. The conf files are unchanged except for the obvious number changes but nothing I can do sets the outgoing caller id. We're using the same version of idefisk and the same version of asterisk (1.2.4-bri stuffed). I found a wiki which said that the DDI numbers we want as caller IDs need to be flagged as allowed CallerID number - this is done by BT - but BT do not seem to understand this. Also our old local exchange was a System X but the new one is System Y. Anyone any ideas on this? Paul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No rings before auto attendant
Thanks, will try this... I actually don't really want to delay incoming calls before the attendant, but it seems to take about 7-10 seconds from the time I dial until the AA picks up, without a ring, it just sounds odd, like the call didn't go through...so I wanted to experiment with trying to add some kind of ringing sound...we'll see if this is actually a good idea or not when I mod this tonight. Thanks all for the tips! Hi all, been searching not finding an answer to this, although I'm guessing it's absurdly simple... I just hooked up a T1 to our * box (1.2.0), which had been using POTS lines via a channel bank.. Now when I call the new T1 circuit, there are no rings, the Autoattendant just picks up right away.. You need to provide audible ring: exten = 2368,1,Ringing exten = 2368,2,Wait(11) exten = 2368,3,Answer and so on. Of course, if you're on a T1, why would you want to artificially delay the calling party's access to the auto-attendant? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK experts only. BT Outgoing caller ID not showing
We had a problem like this until BT enabled callerid (an optional extra) on the line. Julian. Paul Redstone wrote: Hi Just moved offices in the UK and moved our Asterisk box from old one to new one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces. At both offices we had one standard number and a DDI range, routed with Asterisk. We'd set up the configuration so each idefisk set its own caller ID which then got sent by the extensions.conf script. Worked fine at old place but in new place the only number which is received is the central switchboard number. The conf files are unchanged except for the obvious number changes but nothing I can do sets the outgoing caller id. We're using the same version of idefisk and the same version of asterisk (1.2.4-bri stuffed). I found a wiki which said that the DDI numbers we want as caller IDs need to be flagged as allowed CallerID number - this is done by BT - but BT do not seem to understand this. Also our old local exchange was a System X but the new one is System Y. Anyone any ideas on this? Paul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK experts only. BT Outgoing caller ID not showing
On Fri, May 26, 2006 at 04:58:34PM +0100, Paul Redstone wrote: [snip] I found a wiki which said that the DDI numbers we want as caller IDs need to be flagged as allowed CallerID number - this is done by BT - but BT do not seem to understand this. Also our old local exchange was a System X but the new one is System Y. Anyone any ideas on this? Err System Y ? System X is a Marconi switch, I didn't think they made a Y variant, but hey maybe they do. It's almost certainly set-up in the BT end. Usually you can only set your CLI to a number or range of numbers that have been allocated to the line. If BT have set the line to only allow a single number, that's all you'll ever get. Speak to someone knowledgable if you can at BT or report it as a fault. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 301's drop last two digits of dialed number
Hi Jamie, Take a look at the dialstring in your sip.cfg - you'll need to adjust this to match your local dialing plan. Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On 26-May-06, at 2:49 AM, Jamie Heckford wrote: Hi All, Having a rather annoying problem with the Polycom 301 phones, suspect it to be my dialplan. Basically if you lift the receiver off the handset and dial a number, it will not let you dial a number longer than 10 digits (Can see this being acceptable in US, but in UK its a right pain). As soon as the 10th digit is entered, it starts to dial and the number is invalid. If the phone is left on hook and the number is dialed, it works fine when pressing the 'send' key on the handset as it sends the whole number. Can anyone shed any light on this issue? I thought it could be asterisk is trying to Dial to soon so I added a Wait in the dialplan but it didn't seem to work. Kind regards Jamie Heckford Technical Consultant ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] End of migration: adding support for some an alog phones
Nuthin beats an Atlas: http://www.adtran.com/adtranpx/Doc/0/TUA2HMOPDK3KN6S9LM1FH91169/61200305L2-8 A.pdf Telephony Swiss army knife. You can make it do anything. Be prepared to crap your pants when you see the price, though. At that price, I'll keep my dedicated analog line. but thanks for the info. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] UK experts only. BT Outgoing caller ID not showing
On Fri, 26 May 2006, Steve Kennedy wrote: Err System Y ? System X is a Marconi switch, I didn't think they made a Y variant, but hey maybe they do. System Y is/was a common synonym for AXE10. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PCI Problems
Andrew Kohlsmith wrote: On Friday 26 May 2006 11:15, Rich Adamson wrote: Have you dug into the TDM400 far enough to know whether the common complaints are associated with a hardware design issue, TigerJet issue, or driver? (eg, can any of the issues truly be addressed?) My personal opinion is that the TJ320 (the PCI interface chip) is crap. Total, utter, complete, absolute crap. It can work, yes, but it's first and foremost an economy PCI interface. It's like the Plexor PCI interfaces that let you migrate your ISA design to PCI by plugging a PLX9052 between your ISA interface and the PCI interface. It's a cheap way to get your design on to PCI, but it is very inflexible and VERY ... yuck. My THEORY is that Mark started using it because it was easy to use, but when he went to full-scale production and started selling these things he very quickly discovered how different PCI bridges can be and how much trouble the TJ320 can cause for a rapidly-growing hardware company. The Xilinx Spartan II is much better and far more flexible (but much more expensive, too). I don't doubt he has either personally or had is hardware team research alternatives. They're a very busy and very, VERY smart bunch of people. Mark has brought PCI hardware to the public and supports it, which is something I have not done in any kind of volume. I'd love to hear a detailed, technical response about what they've discovered is wrong, what they are going to do, and where their new products are headed, but I'd also like a 50 acre farm and some horses... :-) Those are the same basic conclusions I came to, but I've never designed a pci card so don't have the experience to say I could do it better. ;) I kind of surmised some of that from the fact that the TigerJet has not been used on any digium card after the first couple that were built. R. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK experts only. BT Outgoing caller ID notshowing
There is a system Y, believe it or not it was introduced after system X BT exchange classes: TXS Strowger TXK Crossbar TXE Electronic TXD Digital further sub categorized as System X or System Y System X - GEC Plessey Telecommunications (GPT) System Y - Ericsson AXE10 To determine whether you are on an X or a Y: Dial *#001#. If you get No services are in operation on this line or a list of services, you're System-X. If you get Sorry, you have dialled an invalid service code it's AXE10. Fadge -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy Sent: 26 May 2006 17:10 To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] UK experts only. BT Outgoing caller ID notshowing On Fri, May 26, 2006 at 04:58:34PM +0100, Paul Redstone wrote: [snip] I found a wiki which said that the DDI numbers we want as caller IDs need to be flagged as allowed CallerID number - this is done by BT - but BT do not seem to understand this. Also our old local exchange was a System X but the new one is System Y. Anyone any ideas on this? Err System Y ? System X is a Marconi switch, I didn't think they made a Y variant, but hey maybe they do. It's almost certainly set-up in the BT end. Usually you can only set your CLI to a number or range of numbers that have been allocated to the line. If BT have set the line to only allow a single number, that's all you'll ever get. Speak to someone knowledgable if you can at BT or report it as a fault. Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hint priority and realtime
have you tried to set the priority to -1 for the hints in the db?don't know if it works, but I saw somewhere that the 'hint' priority was actually -1 inside asterisk... maybe this will work with realtime arch.Just a suggestion... 2006/5/26, Damon Estep [EMAIL PROTECTED]: Can someone shed some light on why the 'hint' feature was implemented in the 'priority' field that is purely an integer in the rest of the dialplan? There seems to be a conflict with realtime and the hint priority, in order to put in the hints you would have to change the priority column in the database from int to char and give up some performance (since int indexes better and priority is a parameter in the select)? More importantly, can anyone answer these questions; Can the hint priority by put in mysql realtime? Is there truly an impact to changing the priority datatype to char or varchar? If it can not be put in realtime, can the hint priority exist in the same context statically, and the numbered portion of the dialplan in realtime? (making it "not so real time") Thanks for any info on this. ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy Signals
Hey everyone,
A few employees have noticed some problem here and there when trying to
make outgoing phone calls. After it happens, they try again, and are
able to call through.
The dial plan for outbound calling looks like below. Which I know they
are getting to the Congestion part (which explains the busy) but what I
can't seem to figure out is the cause for why they are getting sent there.
exten = s,1,SetCallerID(${ARG1})
exten = s,2,Wait(2)
exten = s,3,Dial(${TRUNK1}/${ARG2})
exten = s,4,Congestion(10)
exten = s,104,Congestion(10)
The log for a call looked like this
May 26 12:21:08 VERBOSE[6997] logger.c: -- Channel 0/4, span 1 got
hangup request
May 26 12:21:08 VERBOSE[16613] logger.c: -- Zap/4-1 is circuit-busy
May 26 12:21:08 VERBOSE[16613] logger.c: -- Hungup 'Zap/4-1'
May 26 12:21:08 VERBOSE[16613] logger.c: == Everyone is busy/congested
at this time (1:0/1/0)
My question is it asterisk having an issue with the PRI or is the PRI
really reporting the number is busy. I know one case like this I was
calling home, and which when I got through to them, they were not even
on the phone. Are there any tests that I can run on the T1 card in the
server to the PRI? Any suggestions would be helpful.
Kevin
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Re: [Asterisk-Users] IAX2 + port translation
If memory serves me properly what you are showing looks correct. You server is registering to your provider on port 4569 as it should. Their server is seeing you register from 64.26.155.62 and using the prt 14353 which is the port that your firewall has given that outgoing connection. Possibly that the firewall is removing that connection port after some time and your provider cannot get back to your box? Try setting the reregistration time lower than 60 and see if it helps. Hmm, it looks like I have to edit channels/chan_iax2.c to lower the registration timeout - I'm trying 15 seconds, and we'll see if that makes a difference. (You have to override the provider's requested timeout of 60 seconds). Does anyone have any idea what the IP/port PAT pair timeouts are for the Sonicwall TZ170? I see that someone had a similar problem (PAT timeouts, on an unknown device) here: http://lists.digium.com/pipermail/asterisk-dev/2005-February/009341.html - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] PCI Problems
Hi the list ! I share Ethernet card IRQ with my TDM2400 without any trouble here, on an old Intel motherboard and an old PII400 ! This is another proof that sharing IRQ is not necessary an issue. Best Regards, Francois BERGERET, France. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Andrew Kohlsmith Envoyé : vendredi 26 mai 2006 16:37 À : asterisk-users@lists.digium.com Objet : Re: [Asterisk-Users] PCI Problems On Thursday 25 May 2006 16:11, Sean Cook wrote: What could be the other causes? I have exhausted everything I know how to do. PCI sharing explains it (whether or not it is infact the problem). This card shares the BIOS assigned interrupt with the network card... Audio problems can come for a variety of reasons. They are caused by (but not limited to) things such as - IRQ sharing with another device with a shitty driver or poor hardware - Poor/inconsistent PCI bus behaviour and timing - overloaded CPU or poor kernel parameters which cause timing problems - shitty hardware or drivers which can lock out IRQs for a long time - buggy drivers for the TDM or ethernet hardware - bad PCI tuning with setpci or kernel parameters, latency timers especially - other hardware (PCI bus controller, north or south bridge) issues - faulty hardware - poor cabling (either TDM side or ethernet side) IRQ sharing is often blamed for audio problems but the fact of the matter is that IRQ sharing is *NOT* an issue if the hardware that is sharing the IRQ (and the drivers for that hardware) plays nicely and reacts to the IRQ quickly. PCI is DESIGNED to share IRQs. The trouble comes when vendors take old ISA hardware, port it to PCI and/or don't ensure that they not only share IRQs properly but also do not ensure that their drivers check that their hardware caused the IRQ and react to IRQs quickly. There is NOTHING inherently wrong with sharing IRQs. The IRQ handler needs to check the hardware to see if it was their hardware that generated the IRQ and get the hell out if not. A lot of (poor) drivers do NOT do this. The driver either assumes that the IRQ MUST have been generated by the hardware (which can cause a host of weird problems), or the check takes so long that it causes trouble for the card that DID generate the IRQ. Digium's hardware is more sensitive to IRQ sharing trouble than other hardware for two very simple reasons. The first is that the TDM cards have no real buffering. If the data is not taken from the register it will quickly be overwritten by the next block of data. This is analogous to the old 16450 UARTs of yore. They had a receiver shift register and a 1-byte receiver buffer. If you didn't get the data out of the buffer before the next byte had shifted in, the new byte would be transferred to the buffer and you'd get an overrun error. The 16550 replaced the 1-byte receive buffer with a 16-byte FIFO (IIRC) -- you could trigger an IRQ after the FIFO had filled 'x' bytes, and then service the IRQ, retrieving all bytes received in one fell swoop. And if your IRQ service routine got a little delayed it was no big deal because there was room for another byte or two before you started losing data. This allowed the IRQ volume on busy serial applications to be far lower (up to 16x lower) than before, which allowed for better system utilization. Digium's hardware is like the old 16450. There is no FIFO. This was done consciously, and is not necessarily a bad design -- TDM is VERY sensitive to latencies. The more delay you have, the worse things like echo become. Bringing TDM data into the PC is already pretty laggy. Adding more delay with FIFOs isn't necessarily a good thing. (I would argue that having a 16 byte FIFO and triggering the IRQ on the first position would not be a bad thing nor would it introduce any latency, but that's me. I'd change a few things about Digium's hardware, but there is no arguing at their success.) So back to the problem at hand: if there is significant delay between the IRQ and the IRQ service, you lose data. This leads to chirping/clicking and in the case of T1, HDLC/framing errors, dropped links and bouncing D channels (for PRI). The second reason is that Digium's drivers do a LOT of work in the IRQ handler. Essentially they are poor PCI neighbours. In the past (I have not checked this recently) all of the echo cancellation and heavy lifting was done right inside the IRQ handler, with interrupts disabled. This caused their IRQ service time to be lengthy, and until interrupts are enabled again you essentially lock out any other driver from servicing its hardware. (Basically Digium's drivers do to other drivers what Digium's drivers can't stand to have done to it.) Contrast this with Sangoma's drivers, which get the data into system RAM, set a flag (softIRQ?) and then get the hell out of the IRQ context as quickly as possible. Then whenever the CPU
Re: [Asterisk-Users] End of migration: adding support for some analog phones
On 5/26/06, Mimmus [EMAIL PROTECTED] wrote: Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN -- Asterisk -- E1 cable -- Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I use a TDM2400P board together with the actual TE410P? I've had very good results at one of my clients' locations using the following setup: PRI -- Asterisk w/Digium TE406P -- Adtran channel bank -- Fax machine Assuming you have a spare span on your E1 card, it's probably worth it to get a channel bank with 24 FXS ports and put it in your wiring closet. Even if you don't use all the ports now, you'll be a simple cross-connect away from adding an analog station if and when it's needed in the future. -- Strom Carlson http://www.stromcarlson.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Not able to make any calls
Hi Abhijit,
The error message says it all really. You have a context called
'internal' but not 'from-internal'
With thanks,
Tim
- Original Message -
From: Abhijit [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, May 26, 2006 9:46 AM
Subject: [Asterisk-Users] Not able to make any calls
Hi All,
I have registered abhijit for SIP in asterisk Server.
I am able to register my softphone (SJPhone) to the server using the
name abhijit.
But whenever I try to make any calls I am gettinh the following error
message:-
*CLI
-- Registered SIP 'abhijit' at 172.20.28.85 port 5060 expires 120
May 26 07:34:52 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: Cannot
find extension context 'from-internal'
May 26 07:35:23 NOTICE[2761]: pbx.c:1738 pbx_extension_helper: Cannot
find extension context 'from-internal'
my extension.conf is :-
[globals]
VM_PREFIX = *
RINGTIMER = 15
REGTIME = 7:55-17:05
REGDAYS = mon-fri
RECORDEXTEN =
PARKNOTIFY = SIP/200
OUT_2 = IAX2/fwd
OUT_1 = ZAP/g0
OUTPREFIX_2 =
OUTMAXCHANS_2 = 1
OUTCID_2 = mithunafila672648
OPERATOR =
NULL =
IN_OVERRIDE = forcereghours
INCOMING = GRP-1
FAX_RX_EMAIL = [EMAIL PROTECTED]
FAX_RX = system
FAX =
Eabhijit = SIP
E9002 = SIP
E9001 = SIP
E8002 = SIP
E8001 = SIP
DIRECTORY_OPTS =
DIRECTORY = last
DIAL_OUT_1 = 9
DIAL_OUT = 9
DIAL_OPTIONS = tr
DIALOUTIDS = 1/2/
CALLFILENAME =
AFTER_INCOMING =
[ext-did]
include = ext-did-custom
exten = 672648,1,SetVar(FROM_DID=672648) ;
exten = 672648,2,Goto(ext-group,1,1) ;
[ext-group]
include = ext-group-custom
exten = 1,1,Macro(rg-group,30,,200-201);
[ext-local]
include = ext-local-custom
exten = 8001,1,Macro(exten-vm,[EMAIL PROTECTED],8001)
exten = ${VM_PREFIX}8001,1,Macro(vm,8001)
exten = 8002,1,Macro(exten-vm,[EMAIL PROTECTED],8002)
exten = ${VM_PREFIX}8002,1,Macro(vm,8002)
exten = 9001,1,Macro(exten-vm,[EMAIL PROTECTED],9001)
exten = ${VM_PREFIX}9001,1,Macro(vm,9001)
exten = 9002,1,Macro(exten-vm,[EMAIL PROTECTED],9002)
exten = ${VM_PREFIX}9002,1,Macro(vm,9002)
exten = abhijit,1,Macro(exten-vm,[EMAIL PROTECTED],abhijit)
exten = ${VM_PREFIX}abhijit,1,Macro(vm,abhijit)
[outbound-allroutes]
include = outbound-allroutes-custom
include = outrt-001-9_outside
include = outrt-002-outgoingFWD
[outbound-trunks]
include = outbound-trunks-custom
exten = _${DIAL_OUT_1}.,1,Macro(dialout,1,${EXTEN})
[outrt-001-9_outside]
include = outrt-001-9_outside-custom
exten = _9.,1,Macro(dialout-trunk,1,${EXTEN:1})
exten = _9.,2,Macro(outisbusy) ; No available circuits
[outrt-002-outgoingFWD]
include = outrt-002-outgoingFWD-custom
exten = 393,1,Macro(dialout-trunk,2,${EXTEN},)
exten = 393,2,Macro(outisbusy) ; No available circuits
[internal]
exten = 100,1,Dial(SIP/abhijit)
exten = abhijit,1,Echo()
Can anyone please help ..
Abhijit.
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[Asterisk-Users] VOIP equipment trade-up
Hi, I know that this is not the right place, but Im not aware of any alternative. I have some VOIP equipments I would like to trade-up Digium IAXyS101i Used for 2 days VoipSupply page: http://www.voipsupply.com/product_info.php?products_id=772 VoipSupply Price: $90 Linksys WBP54G 802.11G WIFI Dongle Never Used VoipSupply page: http://www.voipsupply.com/product_info.php?products_id=1094 VoipSupply Price: $45 D-Link DI-102 VOIP QOS Adaptor Packet Prioritizer Never Used VoipSupply page: http://www.voipsupply.com/product_info.php?products_id=1168 VoipSupply Price: $80 I would like to trade the three of them with Linksys SPA-942 SIP Phone Dual RJ45, PoE, Inc. AC P/S VoipSupply page: http://www.voipsupply.com/product_info.php?products_id=1334 VoipSupply Price: $190 If interested, kindly email me off the list [EMAIL PROTECTED] Thanks __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] jitterbuffer causes flaky IAX2 incoming connections?
On 25 May 2006, at 20:43, Dr. Michael J. Chudobiak wrote: I've been having problems with incoming IAX2 calls - some work, but a large fraction are answered with dead air or disconnects from my IAX provider. Disabling the jitterbuffer seems to eliminate the problem (so far)! Has anyone else seen this? I'm using 1.2.6, but I'm not sure what my provider is using. A snippet of the a failed incoming call IAX2 debug is attached below (with jitterbuffer on). Note the HANGUP and INVAL codes. - Mike Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00087ms SCall: 00235 DCall: 3 [70.87.18.51:4569] USERNAME: avtech DATE TIME : 2006-05-25 09:26:46 REFRESH : 60 APPARENT ADDRES : IPV4 64.26.155.62:14353 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00087ms SCall: 3 DCall: 00235 [70.87.18.51:4569] Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP Timestamp: 04016ms SCall: 00379 DCall: 0 [64.26.157.230:4569] CAUSE CODE : 0 Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 0 DCall: 00379 [64.26.157.230:4569] steerpike*CLI There isn't quite enough info in that log to tell what is going on. What you have above is part of 2 separate conversations. You have the tail end of a successful registration with 70.87.18.51 and the HANGUP of a call with 64.26.157.230 which your asterisk seems to be confused about. Could you try it again, and make sure you include the NEW message that starts the call which fails ? (assuming that is that there was a NEW !) Thanks. Tim. Tim Panton [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No rings before auto attendant
On 5/26/06, Dan Elder [EMAIL PROTECTED] wrote: Thanks, will try this... I actually don't really want to delay incoming calls before the attendant, but it seems to take about 7-10 seconds from the time I dial until the AA picks up, without a ring, it just sounds odd, like the call didn't go through...so I wanted to experiment with trying to add some kind of ringing sound...we'll see if this is actually a good idea or not when I mod this tonight. Out of random curiosity, is it a channelized T1, or is it a PRI? -- Strom Carlson http://www.stromcarlson.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] large duration calls
On 5/26/06, Francisco Seratti [EMAIL PROTECTED] wrote: Hello mates, im having calls of about 120 o 130 minutes in my accounting DB but these are calls not made by users. I guess my asterisk is not catching some BYE requests and after some timeout it hangs up the call. Is this issue known? Is there a way to trace this problem? Im using Asterisk 1.0.10 on a i686 linux and mysql for the accounting. I have this problem with different protocols, like SIP and Zap. If you need another specs or something, askme. Cheers, Francisco. Well, first off, I'd suggest that you upgrade to the latest stable version...1.0.10 is comparatively ancient. -- Strom Carlson http://www.stromcarlson.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sangoma A200 4 port FXO card suddenly stopped answer on channels 2, 3, 4
I've been using Asterisk 1.2.6 with a 4 port FXO Sangoma A200 card for the past month without many problems (other than the fact that the Sangoma card doesn't disconnect hung up calls immediately, which I posted about in another thread, and has still not been fixed), however, I had a call from one of our clients today and they complained that our phone system kept ringing through when they called. It turns out that the Sangoma card had suddently decided to stop answering on channels 2,3 and 4, so if someone was using channel 1, then no other calls would be picked up. We could, however, make outgoing calls. I tried restarting Asterisk and it didn't make a difference. I then tried restarting the Wanrouter and it started working again. Has anyone else run into this problem? I really don't like the fact that I had to restart the wanrouter for the channels to start behaving normally again. Our phone system is basically useless if it randomly downgrades to a single line. I'm going to speak to Sangoma and see what they have to say about this, but until then, I was hoping someone might have some advice or suggestions for me to try. Thanks, Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users]Asterisk -IP- Siemens HiPath 3750
Hi Josue, benchevWith your guidance, I want to get back to HiPath right now. But I am on the road, so I can get in touch with that system only on Tuesday. But that's really great new Josue, that you can work out the things from those commercial system. I will be back very soon,Thanks again,NguyenOn 5/26/06, Josué Conti [EMAIL PROTECTED] wrote:Hi I work with equipment Siemens Hipath 3000 and HiPath 4000, I can help you, I do not have manuals technician to send, but if to want can help. Already I established connection asterisk( 1.0.9) with Hipath 3750 with a TE110P and a TMS2, functioned 100%. The equipment says between sim.The asterisk uses HiPath 3750, for access the PSTN and when a linking is for a telephone of asterisk, the Hipath directs the digits for asterisk. I wait to have helped. Greetings Josué 2006/5/25, Benchev [EMAIL PROTECTED]: Hi Nguyen ,I haven't got the opportunity to make my project real due to business obstacles, but I still think that it should work. All that follows is a theory, but there are guys on the list thatmight help you with more practical advises. I have stuck with Hipath 3750 and Asterisk + TE110P. I don't have the manual of Hipath 3500 yet (have to buy from local vendor), so I was not sure are these thing possible Scenario: Asterisk|TE110P-TMS2|Hipath 3750 -(16 CO lines) PSTNI had the same idea because I wanted to save on the card side(single span),and usethe Hipath as a channelbank :-) - Is this possible for Asterisk Users call out using CO lines? Some of Siemens guys told me that I need an DISA card for this? Is this true?Most of the time the Siemens guys don't know what is Asterisk. Basically TE110P *is* a DISA since it gives Direct Inward System Access(if this is what they mean by DISA)Below is a threat I found with exactly the same scenario like yours: http://lists.digium.com/pipermail/asterisk-users/2005-April/093761.htmlAnd this proves that the idea must work. - When the call arrived from PSTN through CO line, can it be forwarded to Asterisk? Again, they says that we require the DISA card. As far as anything gets into Asterisk then you are free to do whatever youwant. I don't know what DISA they are talking about? Do they mean S2Mor similar thing(but TMS2 is S2M)?Anyone?Sorry for not being able to help, but hope somebody else would do it.Benchev___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- With best regards,Nguyen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] jitterbuffer causes flaky IAX2 incoming connections?
There isn't quite enough info in that log to tell what is going on. What you have above is part of 2 separate conversations. You have the tail end of a successful registration with 70.87.18.51 and the HANGUP of a call with 64.26.157.230 which your asterisk seems to be confused about. Could you try it again, and make sure you include the NEW message that starts the call which fails ? (assuming that is that there was a NEW !) Tim, There was no NEW. Some IAX2 messages just aren't reaching me, I think. I think that the real problem is a short timeout (maybe 60 seconds?) in my hardware firewall (Sonicwall TZ170) for UDP address:port pairs in the NAT/PAT translation memory. I've hacked the chan_iax2.c code to force a 15 second registration refresh time, instead of 60 seconds, and so far things have worked much better (i.e., the registration is like a keep-alive for the PAT translation pairs). I'll keep the list posted ... - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sangoma A200 4 port FXO card suddenly stopped answer on channels 2, 3, 4
Mike Garey wrote: It turns out that the Sangoma card had suddently decided to stop answering on channels 2,3 and 4, so if someone was using channel 1, then no other calls would be picked up. We could, however, make outgoing calls. I tried restarting Asterisk and it didn't make a difference. I then tried restarting the Wanrouter and it started working again. Has anyone else run into this problem? Do you have the optional echo canceler? The echo canceler on my A20002D died after two months, resulting in erratic one-way audio. Sangoma sent a replacement after I presented my debugging efforts to my vendor (Telephonyware). The replacement works fine. Try re-seating the FXO option card in the main card. The optional echo canceler card can also be unscrewed and re-installed. Anyway, call your vendor about the fall-through problem. The disconnect problems that you mentioned are the same for any FXO card - see http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html.) I use minmessage=5 maxsilence=3 silencethreshold=128 in voicemail.conf. Seems to work reasonably well. - Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pap2 bridging problems
Well NAT isn't the problem here. I just plugged it directly into the Internet, and it still has the same problems, any other ideas. William Piper wrote: Doesn't the Pap2 have a setting for stun? If so, try that set it to stun.fwdnet.net, at least for testing. If you need to use it for a ton of customers, I'd suggest building one so you can manage it yourself... of course if you have money to burn, you could invest in a good SBC. bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Miles Scruggs Sent: Thursday, May 25, 2006 11:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] pap2 bridging problems Well NAT is set to yes, but DMZ etc isn't an option for this pap2 since it needs to be able to roam around different networks. What other options are there? William Piper wrote: Sounds like a NAT issue. Make sure that in your sip.conf you have nat=yes. If that doesn't work, connect to a stun server, or DMZ your ATA, or port forward all needed ports to the ATA's internal IP. FYI, on several applications where I set port forwarding... I needed to set nat=no to get it to work. bp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Miles Scruggs Sent: Thursday, May 25, 2006 11:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] pap2 bridging problems I'm having a real problem with one of my linksys pap2. On outgoing calls the callee will ring, but caller (pap2) will not here it ring When the callee answers, no audio is transmitted either way. Asterisk reports the call connected and bridged correctly. Now the kicker is that sometimes it works and other times it doesn't. I have had the most luck calling land lines, but sometime cell phones will work also. If someone could help me out that would be awesome. Miles ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1559 (20060525) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1.1559 (20060525) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hint priority and realtime
I put my hints in a separate static context, then set the subscribecontext in sip.conf to make subscriptions look at that context for hints. Perhaps that would work for you? -Jason Damon Estep wrote: Can someone shed some light on why the ‘hint’ feature was implemented in the ‘priority’ field that is purely an integer in the rest of the dialplan? There seems to be a conflict with realtime and the hint priority, in order to put in the hints you would have to change the priority column in the database from int to char and give up some performance (since int indexes better and priority is a parameter in the select)? More importantly, can anyone answer these questions; Can the hint priority by put in mysql realtime? Is there truly an impact to changing the priority datatype to char or varchar? If it can not be put in realtime, can the hint priority exist in the same context statically, and the numbered portion of the dialplan in realtime? (making it “not so real time”) Thanks for any info on this. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PAP-2 Conferencing Problems
Stuart Elvish - Dallas Delta Corporation Pty Ltd wrote: Just come across a problem - we have sent out heaps of PAP-2 ATA's and just discovered that when joined in a conference they are choppy on the up leg (so the other users in the conference will hear them with a choppy sound) but the down leg is perfectly fine (so the end user can hear the conference participants perfectly). Is that into a meetme conference? If so I have noticed that you have to change the default RTP Size (on the PAP2 or Sipura) to .20 instead of .30. Don't know why that is but would be very interested in knowing if it fixes your issue. I have tried also with .60 and it is not only choppy, its totally unintelligible. I have tested the same setup with different brands of ATA's and with IP phones and there aren't any problems. I have also tested a couple of different codecs (g729 and ulaw) and the problem seems to still exist. The problem happens when the ATA is both internal and external to the VoIP server network. Does anyone have any suggestions? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andres Technical Support http://www.telesip.net ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE406P - MFC/R2
you can specify the logging level in unicall.conf
logleve=0-255
select a value from 0 to 8. According to unicall.h the levels are:
enum
{
UC_LOG_ERROR= 1,
UC_LOG_WARNING = 2,
UC_LOG_PROTOCOL_ERROR = 3,
UC_LOG_PROTOCOL_WARNING = 4,
UC_LOG_FLOW = 5,
UC_LOG_CAS = 6,
UC_LOG_DEBUG_1 = 7,
UC_LOG_DEBUG_2 = 8
};
I would recommend a value of 4
Regards
On 5/26/06, Fernando Lujan [EMAIL PROTECTED] wrote:
Fernando Lujan wrote:
Steve Underwood wrote:
The problem is almost solved. The card was configured as a T1 interface,
the selled came and jumped it.
Now I have the following problem. When I call from my legacy pbx, appear
a event:
*CLI May 26 12:04:09 WARNING[5215]: chan_unicall.c:627 unicall_report:
MFC/R2 UniCall/31 - 0001 [1/ 1/Idle /Idle ]
May 26 12:04:09 WARNING[5215]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/31 Detected
May 26 12:04:09 WARNING[5215]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/31 Making a new call with CRN 32769
May 26 12:04:09 WARNING[5215]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/31 1101 - [2/ 2/Idle /Idle ]
May 26 12:04:09 WARNING[5215]: chan_unicall.c:2644 handle_uc_event:
Unicall/31 event Detected
May 26 12:04:54 WARNING[5215]: chan_unicall.c:627 unicall_report: MFC/R2
UniCall/31 - 1001 [2/ 2/Seize ack /Seize ack]
But this event, doesn't enter in the context which I configure in the
unicall.conf file. Do I need to change something else?
Thanks again.
Fernando Lujan
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