Re: [Asterisk-Users] LOOKING TO HIRE
Steven Kalcevich wrote: Cant we all just get along :) Preston went trolling, and he landed a boatful. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc
Thierry Wehr wrote: Our ATA286 and 486 using 1.0.6 have all a broken ILBC A++ have you tried switching iLBC frame size: from 20ms to 30 ms? priit ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Konftel
On Thu, 19 May 2005, Dean Collins wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Peter Svensson Sent: Thursday, 19 May 2005 7:55 PM Another and perhaps easier option for wireless konference phones may be http://www.clearone.com/product_service/product_detail.php?prodid=127 and for larger rooms http://www.clearone.com/product_service/product_detail.php?prodid=198 Do you have a price? In Sweden they wireless ones are about $700. I guess for the flexibility you get it may be a good price, especially if you have a larger room so you occasionally need several devices working as one. Any ideas on quality? Have you used one personally? Not the wireless ones, but their wired conference phones work well for us. What is really nice about them is that several phones can either work standalone or be connected together to form a larger system for larger rooms when the need arises. This works as advertised. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoipSupply.com
Just a quick note, if you typically ship to a different address than your credit card billing address, you can file that address with your credit card company. Most cards allow you to have mulitple addresses on file so that your Address Verfication goes through correctly. Not universally true, I'm afraid. Amex France won't do it and Wells Fargo (calif) won't do it. OTH, Paypal will ship an order billed to Wisconsin to Nigeria with a blick. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] why can't my asterisk restart?
Hi. I use Asterisk-CVS HEAD version. When I type a command 'restart gracefully', then asterisk just stop. Messages are : [root ]# asterisk -vvvgc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Not found (No such file or directory) snip messages. Asterisk Ready. *CLI restart gracefully Waiting for inactivity to perform restart... Executing last minute cleanups == Cleaning up OpenH323 channel driver. == Unregistered channel type 'OH323' [2]WrapperAPI::h323_removeall_capabilities: Removing all capabilities. [2]WrapperAPI::h323_removeall_listeners: Removing all listeners. [2]WrapperAPI::h323_removeall_listeners: Removing listener Listener[ip $*:1720] [2]WrapperAPI::h323_end_point_destroy: Destroying endpoint. 0:08.001 Transactor:9d8f198 H225RAS Read error (4): Interrupted system call 0:08.004 Transactor:9d8f198 Trans Ended listener thread on Transport[remote=ip$211.196.70.53:1719 if=ip$192.168.1.151:10001] [1]WrapGatekeeperServer::WrapGatekeeperServer: Destroying gatekeeper. == Destroying musiconhold processes Yuck! Error in buffer handling...: Broken pipe Asterisk cleanly ending (0). Preparing for Asterisk restart... Restarting Asterisk NOW... Hangup [root ]# What or How do I do to correct? Cheers. Kim. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: cisco 7960 question
Matthew Simpson [EMAIL PROTECTED] uttered the following thing: I have a stupid question. How do you remove line presentations on a cisco 7960 ? I have 3 line presentations I don't use anymore and I can't figure out how to remove them. If you're using TFTP for configuration, firstly remove the relevent configuration from the config file. Then you have to go into the phone menus and remove the line configurations. To do this you go into sip configuration, and in each line you then have to blank out the entry for each config option. Ben ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hint with snom 220 - call pick up
Hi, I am trying to use the support for monitoring extension states of a snom 220. In the extensions.conf file I added: exten = 770,hint,SIP/770 It means that when the snom phone boots, it will register it-self to asterisk as a monitoring phone for 770: Asterisk knows that the SIP/770 is monitored. I am using the 3.56q-beta firmware. It is in the DESTINATION option as it said on the tiki of snom in voip-info.org and not in the LINE option as it is said in the snom220 manual that I put the SIP URL of the monitored phones. I can see the used lines. But call pick up is not working: The snom can't pick up a call which has no answer. Did I made something wrong ? Must I do something more ? Please help me. Alexis. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail wav49 format problem
Filepermission error or the mailbox doesn't exist. Check if /cygdrive/e/pbx/voicemail exists and it has the right permissions. (running under cygwin? cheesus..) Daniel Michael Stahl skrev: I have the voicemail format set to wav49 in my voicemail.conf file. When retrieving voicemails, the first message plays back ok - but then Asterisk hangs up and the log shows the following error. Any idea what's up? May 19 12:57:24 VERBOSE[7860]: Asterisk Ready. May 19 13:48:51 WARNING[7860]: Not a wav file 49 May 19 13:48:51 WARNING[7860]: Unable to open fd on /cygdrive/e/pbx/voicemail/default/2460/INBOX/msg.wav May 19 13:48:51 WARNING[7860]: Unable to open /cygdrive/e/pbx/voicemail/default/2460/INBOX/msg (format ulaw): No such file or directory May 19 13:57:03 VERBOSE[7860]: -- Recording the message May 19 13:57:03 VERBOSE[7860]: -- x=0, open writing: /cygdrive/e/pbx/voicemail/default/2460/INBOX/msg format: wav49, 0x10016550 May 19 13:57:18 WARNING[7860]: Failed to write frame May 19 13:59:17 VERBOSE[7860]: -- Recording the message May 19 13:59:17 VERBOSE[7860]: -- x=0, open writing: /cygdrive/e/pbx/voicemail/default/2460/INBOX/msg format: wav49, 0x10040ac8 May 19 13:59:38 WARNING[7860]: Recv error: Interrupted system call May 19 13:59:38 NOTICE[7860]: RTP: Received packet with bad UDP checksum May 19 13:59:45 WARNING[7860]: Failed to write frame May 19 14:01:07 WARNING[7860]: Not a wav file 49 May 19 14:01:07 WARNING[7860]: Unable to open fd on /cygdrive/e/pbx/voicemail/default/2460/INBOX/msg.wav May 19 14:01:07 WARNING[7860]: Unable to open /cygdrive/e/pbx/voicemail/default/2460/INBOX/msg (format ulaw): No such file or directory May 19 14:02:00 WARNING[7860]: Not a wav file 49 May 19 14:02:00 WARNING[7860]: Unable to open fd on /cygdrive/e/pbx/voicemail/default/2460/Old/msg.wav May 19 14:02:00 WARNING[7860]: Unable to open /cygdrive/e/pbx/voicemail/default/2460/Old/msg (format ulaw): No such file or directory May 19 14:03:03 VERBOSE[7860]: -- Recording the message May 19 14:03:03 VERBOSE[7860]: -- x=0, open writing: /cygdrive/e/pbx/voicemail/default/2460/INBOX/msg format: wav49, 0x10040e08 May 19 14:03:16 WARNING[7860]: Failed to write frame May 19 14:03:31 WARNING[7860]: Recv error: Interrupted system call May 19 14:03:58 WARNING[7860]: Not a wav file 49 May 19 14:03:58 WARNING[7860]: Unable to open fd on /cygdrive/e/pbx/voicemail/default/2460/INBOX/msg.wav May 19 14:03:58 WARNING[7860]: Unable to open /cygdrive/e/pbx/voicemail/default/2460/INBOX/msg (format ulaw): No such file or directory Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME Cryptographic Signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Greetings
Rgds, _ Sifis O. Kapassakis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] lookup for extensions on another SIP Proxy
I've got * registered with 50 SIP extensions. There are two another SIP proxies. I'd like to configure following: 1. Call from outside comes on *. * looks up for an extension 2. If no registered extension is on *, then request is forwarded to SIP proxy 1. 3. If client in not found on SIP Proxy 1, then * forwards request to SIP Proxy2 4. If client is not found SIP Proxy 2 congestion tone is generated. What is the best way to do it? I.N. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Offloading all user/peer autentication to SER?
hi is it possible to do all the user authentication on SER and somehow allow calls proxied by SER through asterisk without any direct user/ peer-to-asterisk authentication? roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to create channel of type 'IAX2' (cause 3)
I try to connect to voipjet, but I get always busy, ... it worked
yesterday, ... no changes on my side
-- Executing SetGroup(SIP/615-829b, iax-voipjet) in new stack
-- Executing Dial(SIP/615-829b,
IAX2/[EMAIL PROTECTED]/011886228357765) in new stack
May 20 18:16:26 NOTICE[9733]: app_dial.c:973 dial_exec_full: Unable to
create channel of type 'IAX2' (cause 3)
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup(SIP/615-829b, ) in new stack
== Spawn extension (default, 9011886228357765, 514) exited non-zero on
'SIP/615-829b'
-- Executing Hangup(SIP/615-829b, ) in new stack
== Spawn extension (default, h, 1) exited non-zero on 'SIP/615-829b'
This is the extension.conf for this part:
exten = _9011Z.,413,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}} ;
VoipJet.com NANPA
exten = _9011Z.,414,hangup
exten = _9011Z.,512,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN:${TRUNKMSD}}
Should I remove the hangup line, so that it can go and try NuFone?
How can I improve this?
bye
Ronald
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Re: [Asterisk-Users] cisco 7960 question
Matthew Simpson wrote: I have a stupid question. How do you remove line presentations on a cisco 7960 ? I have 3 line presentations I don't use anymore and I can't figure out how to remove them. Matthew, This is how I remove line 2 from my 7960: # Line 2 appearance line2_name: UNPROVISIONED # Line 2 Registration Authentication line2_authname: UNPROVISIONED # Line 2 Registration Password line2_password: UNPROVISIONED Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IPswitch cannot delete lines double lines
I use an xml editor to remove extensions that no longer exist. You can then update extensions from the server. Chris Mason www.anguillaguide.com Tel: (305) 704-7249 Fax: (815)301-9759 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Thursday, May 19, 2005 11:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IPswitch cannot delete lines double lines I had to cancel Broadvoice, but IPswitch does not like to delete me that line, ... I use instead voipjet, but this one pops up twice, as well as nufone, ... How can I get the name - info into Zap-1 .. Zap-4 (FXS and FXO type)? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID name lookup AGI script
Well I installed this script on to my system (a few hiccups with php 5 but its not erroring anymore). Still not getting any callerid info to pass to my polycom 500 screen. Could it have anything to do with the fact that the number is prepended with a +1 on the screen? Teliax sends the +1NXXNXX on the number. Is that being stripped by the agi script when it queries 411 and google? Or am I just a dumb fart no quite getting what I'm doing? (this is the likely case!) P.S. If anyone has a suggested script to remove the +1 from the number, it would be helpful in other areas as well... Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help please
what i have at present is asterisk installed on fedora core 3, with a dev kit with 1 fxo and 1fxs module and a TE110p card, using the sample files i can dial asterisk using the dev kit and get the asterisk welcome and congratulations message i have got the zaptel.conf conigured as follows fxoks=1fxsks=4 span=1,1,0,ccs,hdb3,crc4bchan=5-19,21-35dchan=20 loadzone=nldefaultzone=nl the zapata.conf as follows [channels]language=nlcontext=defaultusecallerid=yeshidecallerid=nocallwaiting=yescallwaitinguserid=yesthreewaycalling=yesechocancel=yesechocancelwhenbridged=norxgain=0.0txgain=0.0group=1callgroup=1pickupgroup=1immediate=nocallerid=206388230busydetect=nocallprogress=nomusiconhold=default signalling=fxo_kschannel = 1 signalling fxs_kschannel = 4 what i want to be able to do is this have 5incommingnumbers over thepri(e1) lines and also dialing out over whichever line is available on the pri(e1) line. i would like to know 1/ how do you setup your zapata.conf file for 2 cards??? as i have 1 dev kit and 1 te110p card 2/ how do i get asterisk to detect what number has ben dialled by the outside caller and route it to the appropriate extension, if no answer after 10 seconds ring all the other phones and if no answer after that go to voicemail? and allow other users to pick up someone elses phone from their extension. user10031204161092 (full international number)0204161092 (number as dialled from the netherlands)4161092 (number as dialled from amsterdam) user20031204161091 (full international number)0204161091 (number as dialled from the netherlands)4161091 (number as dialled from amsterdam) user30031204161093 (full international number)0204161093 (number as dialled from the netherlands)4161091 (number as dialled from amsterdam) user40031204161094 (full international number)0204161094 (number as dialled from the netherlands)4161091 (number as dialled from amsterdam) 3/ how do i record calls and set them to a file in the format extensionnmrdatetime.mp3 I know its a lot to ask but once i get it up and running i intend on providing a step by step method of how i installed it from installing fdc3 to up and running so will help many others too so please help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK)
Hi Does that mean that I should set up the msn=* and add DDI's to the extensions.conf? I think that BT class the 1+1 as 'auxilary line working'. Rgds -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of asterisk Sent: 18 May 2005 14:28 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK) MSN will only work on 1 ISDN2 line and cannot be spread across 2 ISDN2 lines. From your description I assume you have 2 calls up and the 3rd call fails. This is because you can only have 2 concurrent calls using MSN on ISDN2. You will find you have a different number range for the second ISDN2 If you want to use both ISDN lines for incoming calls with the same number range then you will need to have the lines converted to 1 + 1 Auxiliary working and have the numbers delivered as DDI. Neil -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lee Norvall Sent: 18 May 2005 13:27 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK) Hi I can see what seems to be both devices in use, so I guess it must be down to the capi.conf (below), does this look correct ??? [interfaces] msn=292880 incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886, 292887, 292888, 292889 outgoingmsn=292880 controller=1 softdtmf=1 ;accountcode= context=demo ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 msn=292xxx incomingmsn=292880, 292881, 292882, 292883, 292884, 292885, 292886, 292887, 292888, 292889 outgoingmsn=292880 controller=2 softdtmf=1 ;accountcode= context=demo ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1 ;deflect=12345678 devices=2 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Armin Schindler Sent: 18 May 2005 12:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 2 x Eicon BRI ISDN devices (UK) On Wed, 18 May 2005, Lee Norvall wrote: Hi I have installed 2 x Eicon BRI ISDN cards into a Suse 9.2 server. We can use all 4 lines for out going calls fine, but on incoming we can only use 2. On calling in using the main msn, the 3rd line gives a an engaged signal. I have unplugged 1 of the cards, and the other card takes the 2 calls. I then swapped this around, and this also works fine. But when using both cards, we can only use 2 line in. There are two possibilities: 1) your Telco doesn't send the 3rd call to your other line. You can verify that by using divactrl mlog -c 1 -o (diva_idi module must be leaded) and see if an incoming call is shown. (use -c 2 for the second card) 2) your configuration of chan_capi is not correct and the 3rd call is ignored/rejected. If you don't use DIVA Server cards with CAPI, forget this mail. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi error2
On Fri, 20 May 2005, Altus Snyman wrote:
Good day all
I get chan_capi 0.3.5 and I got the patch but when I try make it gives
I already asked: What patch do you apply?
this error
{standard input}: Assembler messages:
{standard input}:0: Warning: end of file in string; inserted ''
{standard input}:447: Warning: .stabs: missing comma
make: *** [chan_capi.o] Error 2
please help
Do I need a patch for asterisk 1.0.7
No, I have it running here in that configuration.
Armin
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[Asterisk-Users] call barring
Hello, I'm willing to implement call barring for incoming and outgoing calls and I would like to discuss it with the listfirst, since I think It can't be implemented in a 'natural way' and I will need to use agi scripting - database. Process would be: 1. incoming calls priority 1, call incoming.agi, select all the blocked cli's for the called user, if caller is on the list - congestion 2. outgoing calls priority 1, call outgoing.agi, select all the blocked numbers for the caller user, if number dialled is on the list - hang-up or play message. Does It sound like correct? What do you think about execute a script for every call, query the database each time..? Do you think It'll overcharge the system? Thanks a lot and kind regards, Victor. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GOTO statement in Realtime-Extensions not workinglike expected
Use the Goto statement with '|' instead of ','. And make tables for each
context you have in the extensions.conf file.
One thing I noticed is using Goto in real time extensions causes
the jump back to the extensions.conf file.
So first jump to extensions.conf and then specify another switch
statement. But make a new table for each context in the extensions.conf.
Regards:
Bharat M. Sarvan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, May 19, 2005 3:24 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GOTO statement in Realtime-Extensions not
workinglike expected
Hi .. When I use the GoTo statement in realtime to goto a priority only
... E.g. Goto(3) then there's no problem
But, If I try to jump to another context ... E.g.
Goto(othercontext,${EXTEN},3) then it doesn't work
If I process the same statement in extensions.conf things go well
Are there things broken regarding GoTo in combination with Realtime
Extensions ?
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Re: [Asterisk-Users] chan_capi error2
On
http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI
it tells u if u use the cvs as of april you need a patch
I have bot
I tried and it compiled and there is no errors in asterisk startup
What did u change in the capi.conf file?Is it ok if I just change the
context
Thanks
Altus
On Fri, 2005-05-20 at 13:35, Armin Schindler wrote:
On Fri, 20 May 2005, Altus Snyman wrote:
Good day all
I get chan_capi 0.3.5 and I got the patch but when I try make it gives
I already asked: What patch do you apply?
this error
{standard input}: Assembler messages:
{standard input}:0: Warning: end of file in string; inserted ''
{standard input}:447: Warning: .stabs: missing comma
make: *** [chan_capi.o] Error 2
please help
Do I need a patch for asterisk 1.0.7
No, I have it running here in that configuration.
Armin
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Re: [Asterisk-Users] Outbound dialing issue with FXO
However, outbound calls are hit or miss. Sometimes they work fine and other times we get a you must first dial a 1 or 0 message back from telco when dialing out standard POTS lines. Did you get this working yet? Yes, it does seem to be working fine now by adding the ws to the dialstring. The following is intended to provide a little bit of info as to why the w is needed when dialing via some central offices (pstn fxo's). Several telephone companies still have older electro-mechanical central office switches. The majority of these older switches have a bank of dtmf receivers that are shared across all pstn lines, and are only attached to each pstn line during the initial few seconds of a pstn call. (There might be 20 or 30 receivers for a central office switch that supports 5,000 pstn lines.) When asterisk seizes the pstn line (goes off-hook), dial tone is usually provided within a second or two. However, the dtmf receiver may or may not be attached and ready to receive dtmf digits in that short period of time. (If the central office switch is slightly under-engneered, there could also be a shortage of dtmf receivers that _could_ result in a receiver not being attached to the pstn line within the first second or so.) One or more w in the dial string causes asterisk to delay sending the dtmf digits, compensating for the delayed attachment of the dtmf receiver in those central offices. In very very general terms, the delay is only seen when interfacing with analog pstn lines. (Newer central office switches that support isdn typically are not designed/engineered with this one-to-many dtmf receiver arrangement.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie on IVR
Hi, I get fascinated when I dial someone and get an IVR play " for accounts department press 1, for sales, press 2 or hold the line for an operator" and thenhaveMOH play beforethe line is picked up at the desired extesion. Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction. Thanks in anticipation. Mike Discover Yahoo! Get on-the-go sports scores, stock quotes, news & more. Check it out!___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi error2
On Fri, 20 May 2005, Altus Snyman wrote:
On
http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI
it tells u if u use the cvs as of april you need a patch
I have bot
I tried and it compiled and there is no errors in asterisk startup
I don't think the patch is necessary with your version, but it contains a
fix.
I don't know what the problem with your compilation is, maybe you can
provide more output.
What did u change in the capi.conf file?Is it ok if I just change the
context
Sorry, but what do you mean? You need to setup up a capi.conf according to
your ISDN lines/numbers.
Armin
On Fri, 2005-05-20 at 13:35, Armin Schindler wrote:
On Fri, 20 May 2005, Altus Snyman wrote:
Good day all
I get chan_capi 0.3.5 and I got the patch but when I try make it gives
I already asked: What patch do you apply?
this error
{standard input}: Assembler messages:
{standard input}:0: Warning: end of file in string; inserted ''
{standard input}:447: Warning: .stabs: missing comma
make: *** [chan_capi.o] Error 2
please help
Do I need a patch for asterisk 1.0.7
No, I have it running here in that configuration.
Armin
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Re: [Asterisk-Users] call barring
Does It sound like correct? The AGI should set an Variable to indicate if to block or not. What do you think about execute a script for every call, query the database each time..? Do you think It'll overcharge the system? Usually that should not be a problem since it is run at call initiation only. How many call-initiations do you have per second? If it is about one or more you should redesign your AGI: Build it as an native executable use the Fast-Agi-Extension and ommit the DB-Connection (recompile on changed block list). Elmar ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi error2
A fix for what?
I think the patch in that link is broken because I had to take out a lot
of end of lines
Dont you maybe have a working patch
Thanks for the help
Just a question about the conf file
msn and incomingmsn
What is the difference
is msn what you uses when you with the Dial command and incomingmsn is
what is send to extensions.conf?
Thanks again
Altus
On Fri, 2005-05-20 at 14:32, Armin Schindler wrote:
On Fri, 20 May 2005, Altus Snyman wrote:
On
http://www.voip-info.org/wiki-Asterisk+How+to+connect+with+CAPI
it tells u if u use the cvs as of april you need a patch
I have bot
I tried and it compiled and there is no errors in asterisk startup
I don't think the patch is necessary with your version, but it contains a
fix.
I don't know what the problem with your compilation is, maybe you can
provide more output.
What did u change in the capi.conf file?Is it ok if I just change the
context
Sorry, but what do you mean? You need to setup up a capi.conf according to
your ISDN lines/numbers.
Armin
On Fri, 2005-05-20 at 13:35, Armin Schindler wrote:
On Fri, 20 May 2005, Altus Snyman wrote:
Good day all
I get chan_capi 0.3.5 and I got the patch but when I try make it gives
I already asked: What patch do you apply?
this error
{standard input}: Assembler messages:
{standard input}:0: Warning: end of file in string; inserted ''
{standard input}:447: Warning: .stabs: missing comma
make: *** [chan_capi.o] Error 2
please help
Do I need a patch for asterisk 1.0.7
No, I have it running here in that configuration.
Armin
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Re: [Asterisk-Users] Cisco Call Manager Asterisk for Voicemail
3.3.6, so I would either have to use MGCP or H.323. John Riek wrote: What version of Call Manager are you using? __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users . ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GOTO statement in Realtime-Extensions not workinglike expected
I was just going to ask this same question
Is this the normal behavior that you have to do, jump back to the .conf file?
It is how I have it configured, but it's more a hybrid than a true
realtime system.
Thanks
Use the Goto statement with '|' instead of ','. And make tables for each
context you have in the extensions.conf file.
One thing I noticed is using Goto in real time extensions causes
the jump back to the extensions.conf file.
So first jump to extensions.conf and then specify another switch
statement. But make a new table for each context in the extensions.conf.
Regards:
Bharat M. Sarvan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Thursday, May 19, 2005 3:24 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] GOTO statement in Realtime-Extensions not
workinglike expected
Hi .. When I use the GoTo statement in realtime to goto a priority only
... E.g. Goto(3) then there's no problem
But, If I try to jump to another context ... E.g.
Goto(othercontext,${EXTEN},3) then it doesn't work
If I process the same statement in extensions.conf things go well
Are there things broken regarding GoTo in combination with Realtime
Extensions ?
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[Asterisk-Users] ref: Cisco 7960 question
Message: 5 Date: Thu, 19 May 2005 21:44:11 -0500 From: Matthew Simpson [EMAIL PROTECTED] Subject: [Asterisk-Users] cisco 7960 question To: asterisk-users@lists.digium.com I have a stupid question. How do you remove line presentations on a cisco 7960 ? I have 3 line presentations I don't use anymore and I can't figure out how to remove them. If you look in your TFTP folder, you should see a file SIPthe phones mac address.cnf This is the profile of the phone which it requests when it is booting up, so just edit out the details in the relevant sections. I have also seen SIPthe phones mac address.cnf.xml, which will also require editing. Regards Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GOTO statement in Realtime-Extensions notworkinglike expected
Bharat M. Sarvan wrote: Use the Goto statement with '|' instead of ','. And make tables for each context you have in the extensions.conf file. One thing I noticed is using Goto in real time extensions causes the jump back to the extensions.conf file. So first jump to extensions.conf and then specify another switch statement. But make a new table for each context in the extensions.conf. It isn't necessary to make a new table for each context. And if Goto's behavior inside ARA is ture to what you say above, that is a bug and needs to be confirmed and reported. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound dialing issue with FXO
Rich Adamson wrote: However, outbound calls are hit or miss. Sometimes they work fine and other times we get a you must first dial a 1 or 0 message back from telco when dialing out standard POTS lines. Did you get this working yet? Yes, it does seem to be working fine now by adding the ws to the dialstring. The following is intended to provide a little bit of info as to why the w is needed when dialing via some central offices (pstn fxo's). Several telephone companies still have older electro-mechanical central office switches. Perhaps, bu AFAIK, there are NONE left in North America, certainly none left in the US or Canada. The majority of these older switches have a bank of dtmf receivers that are shared across all pstn lines, and are only attached to each pstn line during the initial few seconds of a pstn call. Crossbar systems. Step by step systems converted to DTMF had DTMF to pulse dedicated to each linefinder. (There might be 20 or 30 receivers for a central office switch that supports 5,000 pstn lines.) When asterisk seizes the pstn line (goes off-hook), dial tone is usually provided within a second or two. However, the dtmf receiver may or may not be attached and ready to receive dtmf digits in that short period of time. Not so sure about that. IF dial tone is provided, the receiver is ready and waiting, (If the central office switch is slightly under-engneered, OR overloaded, there could also be a shortage of dtmf receivers that _could_ result in a receiver not being attached to the pstn line within the first second or so.) None of this excuses the inability or unwillingness of Asterisk to listen for Dial Tone. The modem card used for single FXO ( the X100P and clones ) certainly had that ability in its former life as a modem. Has this ever been reported as a bug? Or would this be considered a feature request , along with detection of stutter dial tone on analog lines. One or more w in the dial string causes asterisk to delay sending the dtmf digits, compensating for the delayed attachment of the dtmf receiver in those central offices. It seems to be necessary in electronic offices as well, when dial tone is delayed. John Novack ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoipSupply.com
I just wanted to chime in here..while i have not ever actually ordered anything from voipsupply.com yet, I actually have been to their offices recently. It turns out they are in the same town as I am! I was most impressed. voipsupply.com is not some guy running an internet store out of hise basement/garage. They have a very nice set of offices in a nice office park. They have also been in the hardware space for many years before they even started to deal with VOIP, so this technology and business is not necessarially new to them. I have not ordered stuff from them simply because i did not know they existed before a few months ago, but as i need more VOIP hardware, there is no doubt in my mind that i will be getting everything from them. So i just wanted to give my $.02 here and vouch that they are a legitimate company with what seems to be legitimate and good people. Mark Musone On 5/20/05, Wilson Pickett [EMAIL PROTECTED] wrote: Just a quick note, if you typically ship to a different address than your credit card billing address, you can file that address with your credit card company. Most cards allow you to have mulitple addresses on file so that your Address Verfication goes through correctly. Not universally true, I'm afraid. Amex France won't do it and Wells Fargo (calif) won't do it. OTH, Paypal will ship an order billed to Wisconsin to Nigeria with a blick. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound dialing issue with FXO
Rich Adamson wrote: However, outbound calls are hit or miss. Sometimes they work fine and other times we get a you must first dial a 1 or 0 message back from telco when dialing out standard POTS lines. Did you get this working yet? Yes, it does seem to be working fine now by adding the ws to the dialstring. The following is intended to provide a little bit of info as to why the w is needed when dialing via some central offices (pstn fxo's). Several telephone companies still have older electro-mechanical central office switches. The majority of these older switches have a bank of dtmf receivers that are shared across all pstn lines, and are only attached to each pstn line during the initial few seconds of a pstn call. (There might be 20 or 30 receivers for a central office switch that supports 5,000 pstn lines.) When asterisk seizes the pstn line (goes off-hook), dial tone is usually provided within a second or two. However, the dtmf receiver may or may not be attached and ready to receive dtmf digits in that short period of time. (If the central office switch is slightly under-engneered, there could also be a shortage of dtmf receivers that _could_ result in a receiver not being attached to the pstn line within the first second or so.) One or more w in the dial string causes asterisk to delay sending the dtmf digits, compensating for the delayed attachment of the dtmf receiver in those central offices. In very very general terms, the delay is only seen when interfacing with analog pstn lines. (Newer central office switches that support isdn typically are not designed/engineered with this one-to-many dtmf receiver arrangement.) This makes perfect sense now as the systems we are installing are in small town/rural NC where there is very likely to be some older equipment. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3000 Question
Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out from the cordless phone trough the Sipura - Asterisk link, using the PTSN line on the other port of the Sipura. So far, so good. BUT: While I can receive a phone call arriving on the PSTN port, it is correctly routed to the cordless phone on the other spa port with the faked callerid trick found in the wiki, the spa does not seems to detect the end of the call. So after the other party ends the call, I end up with an open SIP channel on the asterisk server, and what is way worse, the SPA can not accept or dial out any other call on the PSTN line. I have to manually reset it (and restart the asterisk server to get rid of the zombie SIP channel). The point is in other words how to setup the end of a call detection. I assume that the phone line I am using is set up with italian (or european / etsi) standards. How should I setup the end of call detection for this kind of pstn line? Thanks for any help, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom takes long time for reboot to access web page
When I change a setting via the web interface on a polycom 500, it takes minutes to allow access through the web interface again. Any idea why it is so slow? Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] paging thru sipura-841
Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( extextext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RDNIS (DNID) Call Routing
I haven't been able to find much support for the RDNIS or DNID variables online. I am trying to prove a concept of call routing before we move towards development of a production system. I need to have calls routed coming into a call center based on DNIS. What type of syntax is needed in the extensions.conf file and how can I test it with a softphone (ie: can I emulate the DNIS from xlite)? Thanks in Advance. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound dialing issue with FXO
John Novack wrote: ...along with detection of stutter dial tone on analog lines. This would be somewhat useful. When there is stutter for whatever reason, outbound dialing must be delayed a couple seconds. Some times my PSTN line would roll-over to the telco voicemail before * would answer, someone would leave a message, then outbound dialing would fail until the PSTN voicemail was cleared. (Now my * answers on first ring so this isn't a problem anymore.) Would this have to be done in the Zaptel hardware driver, the chan_zap code, or somewhere in the mainline code? I've seen the tone detection routines in dsp.c, could these be brought to bear? (Answer in -dev if this gets off -user subject too much.) -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi error2
On Fri, 20 May 2005, Altus Snyman wrote: A fix for what? A fix with a wrong filedescriptor. Without that fix chan_capi does not work on kernels = 2.6.11 I think the patch in that link is broken because I had to take out a lot of end of lines Dont you maybe have a working patch Thanks for the help Just a question about the conf file msn and incomingmsn What is the difference is msn what you uses when you with the Dial command and incomingmsn is what is send to extensions.conf? Yes. msn defines the own number on dialout and is used to select the controller for dialout. incomingmsn specifies on what number(s) chan_capi should listen on and is used as extension. Armin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 Gateway
Can the H.323 channel be configured as a gateway from another system? Or can it be configured as an endpoint on another system? Or can it only be connected to actual endpoints like phones? If either of the first two are yes, does anybody have a sample h323.config file? The samples that come with the channel are not very clear. Thanks. Peder ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID name lookup AGI script
Which one are you using? If you're using the script found at http://muware.com/asterisk, then the format of the incoming CID doesn't matter -- my script standardizes the CID number to be used for lookup and presents it as 1nxxnxx. -Original Message- From: Chris Coulthurst [mailto:[EMAIL PROTECTED] Sent: Friday, May 20, 2005 5:49 AM To: 'Brian Dingman'; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] CallerID name lookup AGI script Well I installed this script on to my system (a few hiccups with php 5 but its not erroring anymore). Still not getting any callerid info to pass to my polycom 500 screen. Could it have anything to do with the fact that the number is prepended with a +1 on the screen? Teliax sends the +1NXXNXX on the number. Is that being stripped by the agi script when it queries 411 and google? Or am I just a dumb fart no quite getting what I'm doing? (this is the likely case!) P.S. If anyone has a suggested script to remove the +1 from the number, it would be helpful in other areas as well... Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rack Mount Server Recommendations
While full NEBS certainly adds a great deal of expense and we do have some customers insist on it, most are happy to just have -48vdc power. Currently we use a high end commercial inverter which adds about $3000 to the system cost. I would like to be able to do away with the inverter completely. -- Mike - Original Message - From: Francisco A. Lozano [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 19, 2005 5:25 PM Subject: Re: [Asterisk-Users] Rack Mount Server Recommendations -48V DC? you need a NEBS-certified server? really? they're quite expensive... - Original Message - From: Ken Jones [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, May 19, 2005 11:31 PM Subject: Re: [Asterisk-Users] Rack Mount Server Recommendations On Thursday 19 May 2005 2:03 pm, Michael B. Murdock wrote: Is there anywhere (or anyone) who has compiled some recommendations on rack mount servers for Asterisk? We are currently using Dell 2650 and Dell 2850 but are seeing some problems with the raid controllers failing and are now shopping for a suitable alternative. Ideally the server would be 19in rack mount, build with similar quality to the the Dell's, and have a -48VDC power supply option. Oh yeah, and run asterisk like a champ. We've seen too many Dell servers raid controllers fail. Maybe they work fine under Windows but with linux I wouldn't trust them. We've used SuperMicro in the past and have had no problems. -- Ken Jones ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323 build problems
Hi Read README file first. You will get a clue. thanks for you suggestions, but I always read README file, before starting any installation. I've also googled my problem before post here. Thaks again -- .:FaberK:. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie on IVR
Mike-Olumide, Johnson wrote: I get fascinated when I dial someone and get an IVR play for accounts department press 1, for sales, press 2 or hold the line for an operator and then have MOH play before the line is picked up at the desired extesion. You'll find a simple example of how to accomplish this at the voip-info.org website: http://www.voip-info.org/wiki-Asterisk+tips+IVR+menu This website has a large amount of information for using Asterisk, follow some of the See Also links at the bottom to learn more. -Johnathan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk-oh323 build problems
What versions of OpenH323/Pwlib/asterisk-oh323 are you trying to install? OpenH323=1.12.2 Pwlib=1.5.2 asterisk-oh323=0.2 Thank you Michael Michael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone done the Cisco 7960 FW migration path programmatically?
Has anyone out there scripted the rollthrough migration of the Cisco firmware? It would be fantastic if there was an app that would generate a set of templated .CNF and XML files based on the MAC addys entered, then control and present your .BIN images through TFTP. It could then also send the reboot signal too, walking through the oh-so-ridiculous path from 3.2 (which every 7960 I've bought to date seems to ship with) through whichever image you've bought. If my scripting/programming skills weren't so weak I'd try myself. /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100p cards
I have a customer that has 10 analog lines (he can't get digital where he's located). I'm currently using 3 TDM04B cards but I have the damn echo problem all the time what ever setting I do (I followed all the steps I could find in the Wiki and in this forum but none work). I even tried on a different computer to make sure it wasn't a motherboard problem I made a test with 3 X100p cards and I have no echo problem. I was able to find 2 PowerMac 9600 for a 100$ each (they have 6 PCI slots each) so I was wondering how I can setup the 2 servers to be seen as one. I know I can do a iax connection between the 2. This will work fine for incoming calls but for outgoing how I can tell server1 to use the zap channel on the server2 if all the lines are taken in server1? Thanks Martin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A hook flash sent using RTP for telephony signals (RFC2833) does not flash zap channel
I really hope this happens. I'd love to be able to flash my home Zap line. On 5/16/05, Ken Alker [EMAIL PROTECTED] wrote: I just registered ID 0004283 at http://bugs.digium.com for the problem described in subject (found when using a Linksys PAP2-NA). I don't know where the proper forum is to discuss, so I'm hoping anyone interested will read the bug and let me know your thoughts, either at bugs.digium.com, here, or by emailing me directly (or, please suggest another forum that is more appropriate). As an aside, if you know how to make a Cisco 7960 running SIP send a flash command (SIP, RTP, or otherwise), I'd really like to know. /** Ken Alker [EMAIL PROTECTED]ham radio: KA6SDU Impulse Internet Services http://www.impulse.net Santa Barbara, San Luis Obispo, Ventura, Los Angeles, Orange T-3 / T-1 / ADSL / ISDN / 56K / web hosting / wireless / co-lo ***/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Is it something someone said, was it something someone said? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digital Phones
Guys.. I have a question, maybe it's a simple one, maybe not but its puzzling me.. Avaya, Nortel, etc. use digital phones (are they ADSI?) on their PBX's. How can Asterisk take advantage of these phones? For analog, you cn use FXS, but what do you use for this phones? How can you take advantage or current infraestructure without the need to replace all phones? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZAP/DTMF
Title: ZAP/DTMF Hi all! I have got strange problem with DTMF. This is my test setup Phone1(#101) -- (*1) -E1-- (*2) -- Phone2(#201) *1 handles 1XX numbers *2 same for 2XX *1 and *2 have proper routings so phones can call each other *1 has dial number 103 which answers automatiaclly and get keys and tells their voice equvalent *2 has same functionality on autoresponder #203 Test cases: 1) Phone2 calls 203 - local (*) - asterisk detects dtms and run in proper way 2) Phone2 calls 103 - thru ZAP - works well 3) Phone1 calls 103 - local (*) - works well 4) Phone1 calls 203 - thru ZAP - DTMF are not detected ! So it works only in one way. Any ideas? Btw - with debug on ZAP channels case 2 gives on asterisk2 [ TYPE: DTMF (1) SUBCLASS: 1 (49) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] And on asterisk1 [ TYPE: DTMF (1) SUBCLASS: 1 (49) ] [Zap/1-1] In case 4 -- Asterisk1 has [ TYPE: DTMF (1) SUBCLASS: 1 (49) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [Zap/1-1] And NO MESSAGES on asterisk2! Looking forward to get any help! Br, dmitry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: call barring
I was thinking in use realtime asterisk to decide wether to call the agi or not. I mean, add the agi in the first position of the dialplan or delete it for each user. So I can activate - desactivate it and call the script only when necessary. Thanks, Victor. --Message: 18Date: Fri, 20 May 2005 14:39:44 +0200From: Elmar Haneke [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] call barringTo: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comMessage-ID: [EMAIL PROTECTED]Content-Type: text/plain; charset=us-ascii; format=flowed Does It sound like correct? The AGI should set an Variable to indicate if to block or not. What do you think about execute a script for every call, query the database each time..? Do you think It'll overcharge the system?Usually that should not be a problem since it is run at call initiation only.How many call-initiations do you have per second?If it is about one or more you should redesign your AGI: Build it as an native executable use the Fast-Agi-Extension and ommit the DB-Connection (recompile on changed block list).Elmar-- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SecureTelephony
Ok, now thats a gadget i want to have :-) http://www.global-teck.com/english/newproduct.php http://www.global-teck.com/english/telecomproducts.php Anyone knows something similar that would work with asterisk, or any chances getting this to work? Regards, Andreas _ Need more speed? Get Xtra Broadband @ http://jetstream.xtra.co.nz/chm/0,,202853-1000,00.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 3000 Question
I don't know if it is a phone like issue or not, but try the SPA-3000 setup at http://geekgazette.com. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aldo Bergamini Sent: Friday, May 20, 2005 6:35 AM To: Asterisk Users Subject: [Asterisk-Users] Sipura 3000 Question Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out from the cordless phone trough the Sipura - Asterisk link, using the PTSN line on the other port of the Sipura. So far, so good. BUT: While I can receive a phone call arriving on the PSTN port, it is correctly routed to the cordless phone on the other spa port with the faked callerid trick found in the wiki, the spa does not seems to detect the end of the call. So after the other party ends the call, I end up with an open SIP channel on the asterisk server, and what is way worse, the SPA can not accept or dial out any other call on the PSTN line. I have to manually reset it (and restart the asterisk server to get rid of the zombie SIP channel). The point is in other words how to setup the end of a call detection. I assume that the phone line I am using is set up with italian (or european / etsi) standards. How should I setup the end of call detection for this kind of pstn line? Thanks for any help, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sipura 3000 Question
In the advanced options there are a few options for hang-up detection including tone detection, and silence detection. They also have parameters to adjust timing and sensitivy. IIRC, they are not enabled by default. -Original Message- From: Aldo Bergamini [mailto:[EMAIL PROTECTED] Sent: Friday, May 20, 2005 8:35 AM To: Asterisk Users Subject: [Asterisk-Users] Sipura 3000 Question Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out from the cordless phone trough the Sipura - Asterisk link, using the PTSN line on the other port of the Sipura. So far, so good. BUT: While I can receive a phone call arriving on the PSTN port, it is correctly routed to the cordless phone on the other spa port with the faked callerid trick found in the wiki, the spa does not seems to detect the end of the call. So after the other party ends the call, I end up with an open SIP channel on the asterisk server, and what is way worse, the SPA can not accept or dial out any other call on the PSTN line. I have to manually reset it (and restart the asterisk server to get rid of the zombie SIP channel). The point is in other words how to setup the end of a call detection. I assume that the phone line I am using is set up with italian (or european / etsi) standards. How should I setup the end of call detection for this kind of pstn line? Thanks for any help, Aldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How much CPU power needed for asterisk
All, Of course this also depends on the size of the site. For a large site I would certanly feel more comfortable (and its not that expensive) To use a dual Xeon (64bit proc.-3 Ghz) server platform .But just as a general rule Linux and asterisk do not demand much CPU utilization but I would take a risk for a large site with a great deal of traffic.cheers!! Thanks!! Spondon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Thursday, May 19, 2005 11:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] How much CPU power needed for asterisk To my knowledge and experience you don't need much CPU power. I don't have Digium card only Sipura-3000 units and when I compile something on Gentoo (CPU usage goes up to 99% and stays there when compiling), when calls comes in on PSTN line and/or I make a call out over IP at the same time, nobody notices any difference in call quality, echo or any other problems; so I don't think you have to worry about CPU power for Asterisk. You will more likely run out of IRQ assignment for your internal cards than CPU power :-) #Joseph On Tue, 2005-05-17 at 14:01 -0400, Michael Stahl wrote: I'm thinking of placing Asterisk on an itx motherboard in a tiny case. The ITX motherboards top out around 400Mhz PII (in terms of power relative to a desktop). How much CPU would I need for an office of 50 people? How much disk storage for voicemail + OS? (typical / average) The system will have no PCI cards (no Digium FSO/FXO cards) - everything over the LAN connection. Thanks, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Auto Answer BEEP
I've just received a couple of the Grandstream GXP-2000 enterprise phones for evaluation. When a line on the phone is configured for auto answer, it connects silently. Has anyone been successful in havein a beep sound played to alert the user that he has an autoanswer call? Thanks Bill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registering with second SIP service causes error every 2 seconds - what is going on?
I had my asterisk server working fine with FWD as a SIP provider, so I now added a second SIP provider (voctel). The addition to my sip.conf file is almost identical to FWD, however, asterisk now generates lots of debug messages for some strange reason! In particular, the line "# Testing 127.0.0.1 with 172.31.0.0" shows up every two seconds! (See my log below). If I comment out the register and [section] for the new SIP service, the error goes away. While leaving the line in, the VocTel service appears properly when I "SHOW SIP PEERS", "SHOW SIP USERS", and "SHOW SIP REGISTRY". Can anyone explain what is happening? Why are these testing lines appearing every 2 seconds, and how do I stop them? General info My asterisk server's internal address = 172.31.254.2 My asterisk server's internal netmask = 255.255.0.0 FWD server = 209.91.145.154 VocTel server = 69.90.155.70 (new service) sip.conf sections register =xxx:yyy@fwd.pulver.com/8000 register =xxx:yyy@switch-1.voctel.net/9000 [fwd]context=menuexternal; Give them external caller contexttype=friendsecret=yyyusername=xxxfromuser=xxxfromdomain=fwd.pulver.comhost=fwd.pulver.comdtmfmode=inbandnat=yes canreinvite=no;regexten=8000 ; Show FWD as extension 7000insecure=very ; required for incoming FWD calls [voctel]context=menuexternal; Give them external caller contexttype=friendsecret=yyy username=xxxfromuser=xxxfromdomain=switch-1.voctel.nethost=switch-1.voctel.net;defaultip=172.31.254.2dtmfmode=inbandnat=yescanreinvite=no;regexten=9000 insecure=very ; required for incoming calls my message log= May 20 10:01:33 DEBUG[5520]: # Testing 209.91.145.154 with 172.31.0.0May 20 10:01:33 DEBUG[5520]: Target address 209.91.145.154 is not local, substituting externipMay 20 10:01:33 DEBUG[5520]: Scheduled a registration timeout # 15May 20 10:01:33 DEBUG[5520]: # Testing 69.90.155.70 with 172.31.0.0May 20 10:01:33 DEBUG[5520]: Target address 69.90.155.70 is not local, substituting externipMay 20 10:01:33 DEBUG[5520]: Scheduled a registration timeout # 17May 20 10:01:33 VERBOSE[5520]: [chan_agent]May 20 10:01:33 VERBOSE[5520]: [chan_agent] = (Agent Proxy Channel)May 20 10:01:33 VERBOSE[5520]: == Registered channel type 'Agent' (Call Agent Proxy Channel)May 20 10:01:33 VERBOSE[5520]: == Registered application 'AgentLogin'May 20 10:01:33 VERBOSE[5520]: == Registered application 'AgentCallbackLogin'May 20 10:01:33 VERBOSE[5520]: == Registered application 'AgentMonitorOutgoing'May 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/agents.conf': May 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/agents.conf': FoundMay 20 10:01:33 VERBOSE[5520]: [skipping chan_mgcp]May 20 10:01:33 VERBOSE[5520]: [skipping chan_iax2]May 20 10:01:33 VERBOSE[5520]: [skipping chan_iax]May 20 10:01:33 VERBOSE[5520]: [chan_local]May 20 10:01:33 VERBOSE[5520]: [chan_local] = (Local Proxy Channel)May 20 10:01:33 VERBOSE[5520]: == Registered channel type 'Local' (Local Proxy Channel Driver)May 20 10:01:33 VERBOSE[5520]: [skipping chan_skinny]May 20 10:01:33 VERBOSE[5520]: [chan_oss]May 20 10:01:33 VERBOSE[5520]: [chan_oss] = (OSS Console Channel Driver)May 20 10:01:33 VERBOSE[5520]: == Console is full duplexMay 20 10:01:33 VERBOSE[5520]: == Registered channel type 'Console' (OSS Console Channel Driver)May 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/oss.conf': May 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/oss.conf': FoundMay 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/enum.conf': May 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/enum.conf': FoundMay 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/extconfig.conf': May 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/extconfig.conf': FoundMay 20 10:01:33 VERBOSE[5520]: Asterisk Event Logger restartedMay 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/manager.conf': May 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/manager.conf': FoundMay 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/enum.conf': May 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/enum.conf': FoundMay 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/rtp.conf': May 20 10:01:33 VERBOSE[5520]: == Parsing '/asterisk/etc/rtp.conf': FoundMay 20 10:01:33 VERBOSE[5520]: == Parsing '/cygdrive/e/pbx/config/rtp.conf': May 20 10:01:33 VERBOSE[5520]: == Parsing '/cygdrive/e/pbx/config/rtp.conf': FoundMay 20 10:01:33 VERBOSE[5520]: == RTP Allocating from port range 1 - 13000May 20 10:01:33 VERBOSE[5520]: Asterisk Ready.May 20 10:01:33 DEBUG[5520]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL PROTECTED]' Request 102: FoundMay 20 10:01:33 DEBUG[5520]: Setting NAT on RTP to 0May 20 10:01:33 DEBUG[5520]: Setting NAT on VRTP to 0May 20 10:01:33 DEBUG[5520]: # Testing 172.31.250.5 with 172.31.0.0May 20 10:01:33 DEBUG[5520]: Stopping retransmission on
[Asterisk-Users] Raw Hangup 69.73.19.178:4569
Can anyone tell me why I keep getting these messages from IAXTEL? It does appear to register since I get lines like this: 2005-04-30 04:26:42 VERBOSE[1644]: -- Registered to '69.73.19.178', who sees us as 67.182.152.242:4569 But what is this? I dont think IAXTEL is working for me, since I cant dial 800 #s through it when I copy the iaxtel.com instructions. 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:45 DEBUG[18940]: Sending VNAK 2005-05-20 06:55:46 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=908 2005-05-20 06:55:46 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=908 2005-05-20 06:56:27 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=558 2005-05-20 06:56:28 DEBUG[18936]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found 2005-05-20 06:56:37 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=558 2005-05-20 06:56:48 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=558 2005-05-20 06:57:15 DEBUG[18940]: Immediately destroying 7, having received INVAL 2005-05-20 06:57:28 DEBUG[18936]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found 2005-05-20 06:58:07 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=265 2005-05-20 06:58:17 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=265 2005-05-20 06:58:27 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=265 2005-05-20 06:58:28 DEBUG[18936]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found 2005-05-20 06:58:54 DEBUG[18940]: Sending VNAK 2005-05-20 06:58:55 DEBUG[18940]: Sending VNAK 2005-05-20 06:59:05 DEBUG[18940]: Sending VNAK 2005-05-20 06:59:05 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:05 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:05 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:05 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:05 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:06 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:06 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:07 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:07 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:07 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:16 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:16 DEBUG[18940]: Raw Hangup 69.73.19.178:4569, src="" dst=734 2005-05-20 06:59:28 DEBUG[18936]: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found Chris Coulthurst [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: [Asterisk-biz] Asterisk at ISPcon
On Thu, 19 May 2005, Tim Simms wrote: Have a link? http://www.ispcon.com If you are interested in manning the booth, talk to Rick Segrest. -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ToneCommander
Not sure if anyone has played with one of these gems - ToneCommander 7210 -- but they do ISDN over H323 - and seem to be proprietary to Lucent / AG iMerge. I'm trying to find a way to reverse engineer it to work on a standard asterisk setup... The first thing I found with a tcpdump / ethereal is it's trying to communicate udp on 1719.. I assume that I need to run maybe something like gnugk to get it to respond... Anyone been down this road and have any advice? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID name lookup AGI script
On May 20, 2005, at 3:48 AM, Chris Coulthurst wrote:
Well I installed this script on to my system (a few hiccups with php 5
but its not erroring anymore).
Still not getting any callerid info to pass to my polycom 500 screen.
Could it have anything to do with the fact that the number is prepended
with a +1 on the screen? Teliax sends the +1NXXNXX on the
number.
Is that being stripped by the agi script when it queries 411 and
google?
Or am I just a dumb fart no quite getting what I'm doing? (this is the
likely case!)
P.S. If anyone has a suggested script to remove the +1 from the
number,
it would be helpful in other areas as well...
I always conform the number before passing it anywhere. From Teliax I
do this:
exten = s,1,Answer()
exten = s,2,SetCallerID(${CALLERIDNUM:2})
exten = s,3,AGI(callerid.agi|${CALLERIDNUM})
Hope that helps.
/rg
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RE: [Asterisk-Users] CallerID name lookup AGI script
Yeah that's the one that I'm using. The incoming context in extensions.conf has the following entry: exten = MYNUMBER,1,EAGI(cid_rewrite.php,us) exten = MYNUMBER,2,Macro(stdexten,201,SIP/201) Pretty straight forward. The phone still shows just the number when I call inbound. I modified the agi_config.php script to have mysql data proper, and when I run the script from the shell, I can hit enter and go through the steps: [EMAIL PROTECTED] agi-bin]# ./cid_rewrite.php VERBOSE Format CID VERBOSE VERBOSE Formatted Number: VERBOSE Extracted Name: VERBOSE Name: Unknown Caller VERBOSE Number: VERBOSE Address: SET CALLERID Unknown Caller [EMAIL PROTECTED] agi-bin]# Thanks, Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Jay Milk |Sent: Friday, May 20, 2005 7:09 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; 'Brian |Dingman' |Subject: RE: [Asterisk-Users] CallerID name lookup AGI script | |Which one are you using? If you're using the script found at |http://muware.com/asterisk, then the format of the incoming CID doesn't |matter -- my script standardizes the CID number to be used for lookup |and presents it as 1nxxnxx. | | -Original Message- | From: Chris Coulthurst [mailto:[EMAIL PROTECTED] | Sent: Friday, May 20, 2005 5:49 AM | To: 'Brian Dingman'; 'Asterisk Users Mailing List - | Non-Commercial Discussion' | Subject: RE: [Asterisk-Users] CallerID name lookup AGI script | | | Well I installed this script on to my system (a few hiccups | with php 5 but its not erroring anymore). | | Still not getting any callerid info to pass to my polycom 500 screen. | | Could it have anything to do with the fact that the number is | prepended with a +1 on the screen? Teliax sends the | +1NXXNXX on the number. Is that being stripped by the agi | script when it queries 411 and google? | | Or am I just a dumb fart no quite getting what I'm doing? | (this is the likely case!) | | P.S. If anyone has a suggested script to remove the +1 from | the number, it would be helpful in other areas as well... | | Chris Coulthurst | [EMAIL PROTECTED] | | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/aster isk-users | To | UNSUBSCRIBE or update options visit: | |http://lists.digium.com/mailman/listinfo/asterisk-users | | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NVFaxDetect on Gentoo
Hi, I've merged asterisk-0.9.0 on a gentoo (kernel 2.4) system using portage ebuilds. I've just got NVFaxDetect .c files from Justin Newman and I'm about to install them. I want to know which is the best way to accomplish this. Thanks in advance. -- Juan Luis Moyano [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SecureTelephony
The products are basically modems so any setup that would support modems can support these devices. They basically act as point to point modems that send the voice as an encrypted data stream. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andreas Anderson Sent: Friday, May 20, 2005 10:27 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SecureTelephony Ok, now thats a gadget i want to have :-) http://www.global-teck.com/english/newproduct.php http://www.global-teck.com/english/telecomproducts.php Anyone knows something similar that would work with asterisk, or any chances getting this to work? Regards, Andreas _ Need more speed? Get Xtra Broadband @ http://jetstream.xtra.co.nz/chm/0,,202853-1000,00.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Stange question...
Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing #, and she heard a buzz noise in the phone and the phone then SHOCKED her in her ear. She wasn't able to do anything with the phone for a few seconds as the buttons didn't respond, then she could go back to picking up calls and whatnot. This is a Cisco 7960, SIP 7.4 on power over ethernet. I don't see how it would be possible for her to get physically shocked by the phone. Has anyone ever heard of this happening on any type of voip hardware? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID name lookup AGI script
I have been trying to get this script to work as well cid_rewrite However this is what the CLI reports: -- Executing EAGI(Zap/1-1, cid_rewrite/cid_rewrite.php|us) -- Launched AGI Script /var/lib/asterisk/agi-bin/cid_rewrite.php AGI Tx agi_request: cid_rewrite.php AGI Tx agi_channel: Zap/1-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1116600703.79 AGI Tx agi_callerid: 619xxx -- it is getting caller id number AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 0 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: unknown AGI Tx agi_rdnis: unknown AGI Tx agi_context: incoming-zap AGI Tx agi_extension: s AGI Tx agi_priority: 2 AGI Tx agi_enhanced: 1.0 AGI Tx agi_accountcode: AGI Tx -- AGI Script cid_rewrite.php completed, returning 0 -- Executing Dial(Zap/1-1, IAX2/4000|20|rtT) -- Called 4000 -- Call accepted by 129.46.90.210 (format gsm) -- Format for call is gsm However, it does not have any information being returned. I have edited the agi_config.php to point to where the information is. But it just is not getting anything. Help???... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Boosting Shared Internet Bandwidth for Asterisk
--- Andrew Kohlsmith [EMAIL PROTECTED] wrote: You want Quality of Service. Google around, and then look at http://www.mixdown.ca/~andrew/dump/rc.tc. It's what I use and it seems to work very well. Could you please tell me where and how to install it Thanks. __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Boosting Shared Internet Bandwidth for Asterisk
--- Andrew Kohlsmith [EMAIL PROTECTED] wrote: You want Quality of Service. Google around, and then look at http://www.mixdown.ca/~andrew/dump/rc.tc. It's what I use and it seems to work very well. Could you please tell me where and how to install it? Thanks. __ Do you Yahoo!? Make Yahoo! your home page http://www.yahoo.com/r/hs ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Boosting Shared Internet Bandwidth for Asterisk
--- Andrew Kohlsmith [EMAIL PROTECTED] wrote: You want Quality of Service. Google around, and then look at http://www.mixdown.ca/~andrew/dump/rc.tc. It's what I use and it seems to work very well. Thank you Andrew, I am trying to figure out why I can't start it, meanwhile if you ran through this post again, this is what i am getting when I do rc.tc start: ./rc.tc start RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists We have an error talking to the kernel RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists RTNETLINK answers: File exists We have an error talking to the kernel iptables v1.2.9: Couldn't load match `p2p':/lib/iptables/libipt_p2p.so: cannot open shared object file: No such file or directory Try `iptables -h' or 'iptables --help' for more information. iptables v1.2.9: Couldn't load match `ipp2p':/lib/iptables/libipt_ipp2p.so: cannot open shared object file: No such file or directory Try `iptables -h' or 'iptables --help' for more information. iptables: No chain/target/match by that name Any suggestions? __ Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie on IVR
Where's the question? -Original Message- From: Mike-Olumide, Johnson [mailto:[EMAIL PROTECTED] Sent: Friday, May 20, 2005 7:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Newbie on IVR Hi, I get fascinated when I dial someone and get an IVR play for accounts department press 1, for sales, press 2 or hold the line for an operator and then have MOH play before the line is picked up at the desired extesion. Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction. Thanks in anticipation. Mike Discover Yahoo! Get on-the-go sports scores, stock quotes, news more. Check it out! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SecureTelephony
Lol, why bother, just carry your laptop and run a voip vpn connection. Btw this part of the web site Customers Outside USA: Snapcell(tm) PC-300 is approved for commercial export by the U.S. Department of Commerce. Means it's crap ha ha :) Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andreas Anderson Sent: Friday, 20 May 2005 10:27 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SecureTelephony Ok, now thats a gadget i want to have :-) http://www.global-teck.com/english/newproduct.php http://www.global-teck.com/english/telecomproducts.php Anyone knows something similar that would work with asterisk, or any chances getting this to work? Regards, Andreas _ Need more speed? Get Xtra Broadband @ http://jetstream.xtra.co.nz/chm/0,,202853-1000,00.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digital Phones
Hi Anton, http://www.abptech.com/mainpages/products/citelGateway.html Have a look at these guys. They do have a gateway for Avaya and Nortel and they say, it is certified with Asterisk. Francois Lambert COO Aheeva Technology Inc. Tel. : 514-223-2581 #2200 Cel. : 514-570-4797 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk RealTime asterisk configuration files through DBMS
Hi Im a new user with Asterisk Realtime. Im trying to write a web application in PHP that will show me asterisk *conf files configuration and that can help me to add/remove users/peers etc... I know that from asterisk 1.0.7 on (with cvs upload) asterisk configuration files can be managed trough aDBMS in my case Mysql. Id want to know if someone knows how can I map configuration files in databases tables: in example if I have a sip conf how can I create a table in my database whose contents are informations stored in sip.conf? Have I to manually insert all my peers in my databases table? Any help of any kind in this argument is well accepted! Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom takes long time for reboot to access web page
Hi Chris - When I change a setting via the web interface on a polycom 500, it takes minutes to allow access through the web interface again. Any idea why it is so slow? I've always had the same thing happen across all versions of the firmware since 1.3.0. The phone will boot and be able to take and make calls, but the web interface won't be accessible for 2-3 minutes after the phone appears completely booted up. I guess the web server just takes a while to get fully loaded from the flash memory. I gave up on the web interface anyway. You just can't do as much with it, and when you have a boatload of handsets it takes way too long to configure them all via the web interface. Hooray for FTP configs. - Noah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Alsa and lag
On Thu, 19 May 2005 18:11:47 -0700, Tyler Spivey [EMAIL PROTECTED] wrote: My problem is this. I am connected via fwd's iax protocol. I try to call the echo test, sometimes I get an answer, sometimes not. When I do, sound is about one second lagged - I still hear the ring about a second after I get the answering on console message, and the echo test takes about a second or second and a half to get back the answer. When I use asterisk with dmix and dsnoop, calling just produces the killed. message, and asterisk exits. I'm just looking for a good quality softphone for the console. Linphone doesn't seem to work properly, and is kludgy. Can someone help me fix my alsa setup or the alsa driver? I'm using asterisk CVS. You could probably use the tkphone that comes with iaxclient. ISTR that it runs in the background, and uses a tcl interface to control it. I would think that you could control it from the console. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stange question...
On May 20, 2005, at 8:08 AM, Mark Johnson wrote: Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing #, and she heard a buzz noise in the phone and the phone then SHOCKED her in her ear. She wasn't able to do anything with the phone for a few seconds as the buttons didn't respond, then she could go back to picking up calls and whatnot. This is a Cisco 7960, SIP 7.4 on power over ethernet. I don't see how it would be possible for her to get physically shocked by the phone. Has anyone ever heard of this happening on any type of voip hardware? Mark What kind of clothing was she wearing? (Static electricity and plastic phones, you know?) Hope she's being nice about it, btw. I've had employees (try to) sue me for less ;-) /rg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_meetme2.so does not load due to KRB5 symbol.
I have just tried to set up the Meetme2 application but I am getting a an asterisk loader error for a KRB5_sname_to_principal symbol when astersik tries to load the app_meetme2.so module. I am using Asterisk on a Redhat 7.3 system. I think the problem is coming from the requirement of Meetme on Postgres - Is it possible to install and compile Meetme2 without Postgres as I am using Mysql anyway. John Melody SyberNet Ltd. Galway Business Park, Dangan, Galway. Tel. No. +353 91 514400 Fax. NO. +353 91 514409 Mobile - 087-2345847 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re:Digital Phones
Anton, Nortel and Avaya are not ADSI phones. I can only see two options run * on ATA port on PBX or buy new phones. I have seen dialogic cards that act like a nortel digital ext but there are no * Drivers as far has i know. My two cents Tom B -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. Thank You For Choosing Cache Communications ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Boosting Shared Internet Bandwidth for Asterisk
On May 20, 2005, at 8:11 AM, chawki hammoud wrote: --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: You want Quality of Service. Google around, and then look at http://www.mixdown.ca/~andrew/dump/rc.tc. It's what I use and it seems to work very well. Could you please tell me where and how to install it Thanks. GOOGLE. LEARN. DEPLOY. You need a primer in IP networking before you endeavor to play with packet shaping or you'll be stabbing in the dark. You also need to ascertain whether or not it will be a complete waste of time if/when your provider completely ingores QoS. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stange question...
Yup. I even suspected it was a 7960 before I got that far in your email. It hasn't happened to any of my users, but I heard about it at a Cisco users group meeting, from a number of people representing a different companies. Cisco was present and stumped, I have heard any more about it though. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Johnson Sent: Friday, May 20, 2005 8:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Stange question... Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing #, and she heard a buzz noise in the phone and the phone then SHOCKED her in her ear. She wasn't able to do anything with the phone for a few seconds as the buttons didn't respond, then she could go back to picking up calls and whatnot. This is a Cisco 7960, SIP 7.4 on power over ethernet. I don't see how it would be possible for her to get physically shocked by the phone. Has anyone ever heard of this happening on any type of voip hardware? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom takes long time for reboot to access web page
I have the same problem with the IP300, 500, and 600s. I think it's just because the phone takes a while to start the web services back up after it reboots. -- Tom Hayden On 5/20/05, Chris Mason (Lists) [EMAIL PROTECTED] wrote: When I change a setting via the web interface on a polycom 500, it takes minutes to allow access through the web interface again. Any idea why it is so slow? Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call disconnects after 30 minutes
We have been experiencing a strange problem lately. Calls are disconnected after approximately 30 minutes when dialing out through a T100p. Asterisk is disconnecting the calls and logging the following: full:May 12 14:03:35 DEBUG[1970]: Bridge stops because we're zombie or need a soft hangup: c0=SIP/sipuser-a58d, c1=Zap/9-1, flags: No,No,No,Yes Asterisk version: 1.0.7 My Zaptel.conf: loadzone = us defaultzone=us loadzone=us defaultzone=us span=1,1,0,esf,b8zs bchan=1-9 dchan=24 I have google'd and wiki'd but cannot seem to find any info on this problem. Anybody else had this problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoipSupply.com
Thank You Cory, I have already touched base with Garrett. -Original Message- From: Cory Andrews [mailto:[EMAIL PROTECTED] Sent: Thursday, May 19, 2005 7:32 PM To: Tracy Phillips; Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoipSupply.com Manjit - if you still have equipment needs we are happy to assist. I apologize that you did not receive a timely response to your original inquiry from our sales department. Email inquiries are typically answered within 12 hours, if not sooner. Like many companies involved in varying aspects of the VOIP industry, we are enjoying tremendous growth and tremendous interest from potential customers. We have been proactively adding additional sales personnel in order to maintain quality service and insure speedy and response times. Please contact [EMAIL PROTECTED] with a brief summary of your needs. I have copied my sales team on this post, and I am certain that you will receive the information you are looking for. Best Regards, Cory Andrews ++ 454 Sonwil Drive Buffalo, NY 14225 ++ v - 800.398.VOIP Ext 22 f - 716.630.1548 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tracy Phillips Sent: Thursday, May 19, 2005 9:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] VoipSupply.com On 5/17/05, Manjit Riat [EMAIL PROTECTED] wrote: I am going to buy some IP phones from them but I sent them an email couple of weeks ago and got no reply. Has anyone ordered anything from them? Any other places that I can buy from? Sorry if it's a wrong post. I would give them a call. I just ordered a 7960 last Friday and it got here on Tuesday. Darren Hartman was great to work with and I highly recommend him: Darren Hartman B2 Technologies, llc www.b2llc.com [EMAIL PROTECTED] AOL IM - darrenb2llc 454 Sonwil Drive Buffalo,NY 14225 (716) 250-3417 x24 office (716) 830-2412 cell (716) 630-1548 fax Have a great day! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trouble getting a SIP phone to dial out through TE100P
We have a test asterisk box setup and can call each other on our sip phones and receive calls in on the PRI to our phones no problem. Our problem is getting asterisk to allow us to dial out using our PRI. Digium has instructed us that we need to strip the leading 9 from the digits dialed. Does anyone having a config that I might glean some lines from to see where we are going wrong. Thanks in advance, Nick Crocker Network Administrator Tri-lakes Internet, Inc. 517 S. Second St. Branson, MO. 65616 [EMAIL PROTECTED] http://www.tri-lakes.net tel: fax: 417-335-7889 417-339-9158 Want a signature like this? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 image001.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Valet Parking and SuperValet Parking - back level
It seems Valet Parking and Super Valet Parking won't compile with CVS after 3/4/2005. Apparently some major reworks to Asterisk has broken the out-of-tree modules. Ref: http://bugs.digium.com/view.php?id=3573 The standard Parking lacks functionality and doesn't support contexts.Yet the more current CVS supports several much needed features like Chan Spy and native MOH as well as numerous bug fixes compared to 1.0.7. As these out-of-tree modules don't qualify for Bug Tracker, I thought it appropriate to open an item on the list. Any insight from those involved developers or suggestions at large are appreciated. Jon Bebeau ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stange question...
I funny one is when our IT manager accidentally supplies the power supply power into one of our voip phones and also feeds it the POE =) Melted outlet, flames in the RJ connector and melted cat5 cable -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dan Austin Sent: Friday, May 20, 2005 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Stange question... Yup. I even suspected it was a 7960 before I got that far in your email. It hasn't happened to any of my users, but I heard about it at a Cisco users group meeting, from a number of people representing a different companies. Cisco was present and stumped, I have heard any more about it though. Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Johnson Sent: Friday, May 20, 2005 8:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Stange question... Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing #, and she heard a buzz noise in the phone and the phone then SHOCKED her in her ear. She wasn't able to do anything with the phone for a few seconds as the buttons didn't respond, then she could go back to picking up calls and whatnot. This is a Cisco 7960, SIP 7.4 on power over ethernet. I don't see how it would be possible for her to get physically shocked by the phone. Has anyone ever heard of this happening on any type of voip hardware? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Stange question...
Are you using POE from a 3550? We have had similar problems, upgrading the firmware on the switch has reduced the occurrences. The Cisco phones are not always nice in an environment with a lot of static electricity. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Johnson Sent: Friday, May 20, 2005 9:09 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Stange question... Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing #, and she heard a buzz noise in the phone and the phone then SHOCKED her in her ear. She wasn't able to do anything with the phone for a few seconds as the buttons didn't respond, then she could go back to picking up calls and whatnot. This is a Cisco 7960, SIP 7.4 on power over ethernet. I don't see how it would be possible for her to get physically shocked by the phone. Has anyone ever heard of this happening on any type of voip hardware? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stange question...
Dan Austin wrote: Yup. I even suspected it was a 7960 before I got that far in your email. It hasn't happened to any of my users, but I heard about it at a Cisco users group meeting, from a number of people representing a different companies. Cisco was present and stumped, I have heard any more about it though. Dan This is interesting. I thought she had fallen off her rocker because she said the one today actually hurt, where the one before she couldn't tell if she got shocked or not. And to answer the last response, she is being nice about it, but I think I'm going to switch out her phone before it happens again!! Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trouble getting a SIP phone to dial out through TE100P
We have a test asterisk box setup and can call each other on our sip phones and receive calls in on the PRI to our phones no problem. Our problem is getting asterisk to allow us to dial out using our PRI. Digium has instructed us that we need to strip the leading 9 from the digits dialed. Does anyone having a config that I might glean some lines from to see where we are going wrong. Thanks in advance, Nick Crocker Network Administrator Tri-lakes Internet, Inc. 517 S. Second St. Branson, MO. 65616 [EMAIL PROTECTED] http://www.tri-lakes.net tel: fax: 417-335-7889 417-339-9158 Want a signature like this? -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.322 / Virus Database: 266.11.14 - Release Date: 5/20/2005 image001.jpg___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stange question...
Mmm i experienced something similar with a grandstream, but i would not say it was a shock, just a strong noise, and i felt my ear like if it were going to explode, but not sure how to avoid it. May be some one here in the list can show us why this happens. On 5/20/05, Mark Johnson [EMAIL PROTECTED] wrote: Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing #, and she heard a buzz noise in the phone and the phone then SHOCKED her in her ear. She wasn't able to do anything with the phone for a few seconds as the buttons didn't respond, then she could go back to picking up calls and whatnot. This is a Cisco 7960, SIP 7.4 on power over ethernet. I don't see how it would be possible for her to get physically shocked by the phone. Has anyone ever heard of this happening on any type of voip hardware? Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Newbie on IVR
Need help to setup IVR with MOH. Thanks for asking, I appreciate your response. Regards, Mike - Original Message - From: Jay Milk [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Friday, May 20, 2005 4:41 PM Subject: RE: [Asterisk-Users] Newbie on IVR Where's the question? -Original Message- From: Mike-Olumide, Johnson [mailto:[EMAIL PROTECTED] Sent: Friday, May 20, 2005 7:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Newbie on IVR Hi, I get fascinated when I dial someone and get an IVR play for accounts department press 1, for sales, press 2 or hold the line for an operator and then have MOH play before the line is picked up at the desired extesion. Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction. Thanks in anticipation. Mike Discover Yahoo! Get on-the-go sports scores, stock quotes, news more. Check it out! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to get in touch with sixTel?
If anybody here is a sixTel customer, can you share any tips tricks for getting in touch with anybody there? They are absurdly hard to get a hold of, particularly when you have a technical issue needing to be resolved. If anyone has any phone numbers other than their main 800 line, I'd sure appreciate it. Thank you, Bryan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stange question...
Eric Alexander wrote: Are you using POE from a 3550? We have had similar problems, upgrading the firmware on the switch has reduced the occurrences. The Cisco phones are not always nice in an environment with a lot of static electricity. POE is coming from a 3500XL I think. It just weird that this has never happened until I changed from Call Manager to Asterisk. I know this has to be a hardware issue but they are blaiming it on Asterisk... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID name lookup AGI script
sorry, not sure if i am understanding your purpose, you want to alter the callerid_id?? if im right, then, could you show me the code inside cid_rewrite.php?? best regards On 5/20/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have been trying to get this script to work as well cid_rewrite However this is what the CLI reports: -- Executing EAGI(Zap/1-1, cid_rewrite/cid_rewrite.php|us) -- Launched AGI Script /var/lib/asterisk/agi-bin/cid_rewrite.php AGI Tx agi_request: cid_rewrite.php AGI Tx agi_channel: Zap/1-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1116600703.79 AGI Tx agi_callerid: 619xxx -- it is getting caller id number AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 0 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: unknown AGI Tx agi_rdnis: unknown AGI Tx agi_context: incoming-zap AGI Tx agi_extension: s AGI Tx agi_priority: 2 AGI Tx agi_enhanced: 1.0 AGI Tx agi_accountcode: AGI Tx -- AGI Script cid_rewrite.php completed, returning 0 -- Executing Dial(Zap/1-1, IAX2/4000|20|rtT) -- Called 4000 -- Call accepted by 129.46.90.210 (format gsm) -- Format for call is gsm However, it does not have any information being returned. I have edited the agi_config.php to point to where the information is. But it just is not getting anything. Help???... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Newbie on IVR
This should help. http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN654 --- Jay Milk [EMAIL PROTECTED] wrote: Where's the question? -Original Message- From: Mike-Olumide, Johnson [mailto:[EMAIL PROTECTED] Sent: Friday, May 20, 2005 7:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Newbie on IVR Hi, I get fascinated when I dial someone and get an IVR play for accounts department press 1, for sales, press 2 or hold the line for an operator and then have MOH play before the line is picked up at the desired extesion. Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction. Thanks in anticipation. Mike Discover Yahoo! Get on-the-go sports scores, stock quotes, news more. Check it out! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Julius Igugu SouthWork Co. Ltd. __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID name lookup AGI script
Run ./cid_rewrite.php from the the shell to see where it's failing. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Friday, May 20, 2005 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] CallerID name lookup AGI script I have been trying to get this script to work as well cid_rewrite However this is what the CLI reports: -- Executing EAGI(Zap/1-1, cid_rewrite/cid_rewrite.php|us) -- Launched AGI Script /var/lib/asterisk/agi-bin/cid_rewrite.php AGI Tx agi_request: cid_rewrite.php AGI Tx agi_channel: Zap/1-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1116600703.79 AGI Tx agi_callerid: 619xxx -- it is getting caller id number AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 0 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: unknown AGI Tx agi_rdnis: unknown AGI Tx agi_context: incoming-zap AGI Tx agi_extension: s AGI Tx agi_priority: 2 AGI Tx agi_enhanced: 1.0 AGI Tx agi_accountcode: AGI Tx -- AGI Script cid_rewrite.php completed, returning 0 -- Executing Dial(Zap/1-1, IAX2/4000|20|rtT) -- Called 4000 -- Call accepted by 129.46.90.210 (format gsm) -- Format for call is gsm However, it does not have any information being returned. I have edited the agi_config.php to point to where the information is. But it just is not getting anything. Help???... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Stange question...
At 10:08 AM 5/20/2005, you wrote: Ok, guys... Please be gentle with me. I have what is going to be the strangest question you will have ever heard, but I have no idea what to tell this person. I set up Asterisk 3 or 4 weeks ago, everything is running smooth. My receptionist has told me on two different occasions that she tried to transfer a call by pressing #, and she heard a buzz noise in the phone and the phone then SHOCKED her in her ear. She wasn't able to do anything with the phone for a few seconds as the buttons didn't respond, then she could go back to picking up calls and whatnot. This is a Cisco 7960, SIP 7.4 on power over ethernet. I don't see how it would be possible for her to get physically shocked by the phone. Has anyone ever heard of this happening on any type of voip hardware? Is she using a headset or a handset with the phone? Also, what kind of floor or floor matt under her chair? Tom Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID name lookup AGI script
What do you see on the * console for incoming calls? There should be Formatted Number and Extracted Name logs. -Original Message- From: Chris Coulthurst [mailto:[EMAIL PROTECTED] Sent: Friday, May 20, 2005 10:35 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] CallerID name lookup AGI script Yeah that's the one that I'm using. The incoming context in extensions.conf has the following entry: exten = MYNUMBER,1,EAGI(cid_rewrite.php,us) exten = MYNUMBER,2,Macro(stdexten,201,SIP/201) Pretty straight forward. The phone still shows just the number when I call inbound. I modified the agi_config.php script to have mysql data proper, and when I run the script from the shell, I can hit enter and go through the steps: [EMAIL PROTECTED] agi-bin]# ./cid_rewrite.php VERBOSE Format CID VERBOSE VERBOSE Formatted Number: VERBOSE Extracted Name: VERBOSE Name: Unknown Caller VERBOSE Number: VERBOSE Address: SET CALLERID Unknown Caller [EMAIL PROTECTED] agi-bin]# Thanks, Chris Coulthurst [EMAIL PROTECTED] |-Original Message- |From: [EMAIL PROTECTED] [mailto:asterisk-users- |[EMAIL PROTECTED] On Behalf Of Jay Milk |Sent: Friday, May 20, 2005 7:09 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion'; 'Brian |Dingman' |Subject: RE: [Asterisk-Users] CallerID name lookup AGI script | |Which one are you using? If you're using the script found at |http://muware.com/asterisk, then the format of the incoming CID doesn't |matter -- my script standardizes the CID number to be used for lookup |and presents it as 1nxxnxx. | | -Original Message- | From: Chris Coulthurst [mailto:[EMAIL PROTECTED] | Sent: Friday, May 20, 2005 5:49 AM | To: 'Brian Dingman'; 'Asterisk Users Mailing List - Non-Commercial | Discussion' | Subject: RE: [Asterisk-Users] CallerID name lookup AGI script | | | Well I installed this script on to my system (a few hiccups with php | 5 but its not erroring anymore). | | Still not getting any callerid info to pass to my polycom 500 screen. | | Could it have anything to do with the fact that the number is | prepended with a +1 on the screen? Teliax sends the | +1NXXNXX on the number. Is that being stripped by the agi | script when it queries 411 and google? | | Or am I just a dumb fart no quite getting what I'm doing? (this is | the likely case!) | | P.S. If anyone has a suggested script to remove the +1 from the | number, it would be helpful in other areas as well... | | Chris Coulthurst | [EMAIL PROTECTED] | | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/aster isk-users To | UNSUBSCRIBE or update options visit: | |http://lists.digium.com/mailman/listinfo/asterisk-users | | | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID name lookup AGI script
No, the script will - conform incoming CID numbers to the 1+10 dialing convention. This allows for easy call-return. - attempt to find a name for this number by doing a reverse-lookup on Google and 411.com. You can download the script at http://muware.com/asterisk It currently requires PHP 5, but I'll make it backward compatible once I have some time. -Original Message- From: Moises Silva [mailto:[EMAIL PROTECTED] Sent: Friday, May 20, 2005 12:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID name lookup AGI script sorry, not sure if i am understanding your purpose, you want to alter the callerid_id?? if im right, then, could you show me the code inside cid_rewrite.php?? best regards On 5/20/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I have been trying to get this script to work as well cid_rewrite However this is what the CLI reports: -- Executing EAGI(Zap/1-1, cid_rewrite/cid_rewrite.php|us) -- Launched AGI Script /var/lib/asterisk/agi-bin/cid_rewrite.php AGI Tx agi_request: cid_rewrite.php AGI Tx agi_channel: Zap/1-1 AGI Tx agi_language: en AGI Tx agi_type: Zap AGI Tx agi_uniqueid: 1116600703.79 AGI Tx agi_callerid: 619xxx -- it is getting caller id number AGI Tx agi_calleridname: unknown AGI Tx agi_callingpres: 0 AGI Tx agi_callingani2: 0 AGI Tx agi_callington: 0 AGI Tx agi_callingtns: 0 AGI Tx agi_dnid: unknown AGI Tx agi_rdnis: unknown AGI Tx agi_context: incoming-zap AGI Tx agi_extension: s AGI Tx agi_priority: 2 AGI Tx agi_enhanced: 1.0 AGI Tx agi_accountcode: AGI Tx -- AGI Script cid_rewrite.php completed, returning 0 -- Executing Dial(Zap/1-1, IAX2/4000|20|rtT) -- Called 4000 -- Call accepted by 129.46.90.210 (format gsm) -- Format for call is gsm However, it does not have any information being returned. I have edited the agi_config.php to point to where the information is. But it just is not getting anything. Help???... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CallerID name lookup AGI script
Hi there, I am trying to the the cid_rewrite.php script but if I run it from the directly I get this error: ./cid_rewrite.php br / bParse error/b: parse error, expecting `T_OLD_FUNCTION' or `T_FUNCTION' or `T_VAR' or `'}'' in b/var/lib/asterisk /agi-bin/astlib_jm.php/b on line b73/bbr / br / bFatal error/b: Cannot instantiate non-existent class: agi in b/var/lib/asterisk/agi-bin/cid_rewrite.php/b on line b60/bbr / Here is the PHP version I am using: PHP 5.0.4 (cgi) (built: May 20 2005 14:08:40) Copyright (c) 1997-2004 The PHP Group Zend Engine v2.0.4-dev, Copyright (c) 1998-2004 Zend Technologies Does anybody know what my problem is or if I am missing anythiong here? Thanks!! Oswaldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
