Re: [asterisk-users] sip forward behind a nat
Yep make the server with dynamic ip register to server with static ip ( sip or iax both will do but in sip keep nat=yes while making extension ) On 12/11/06,
Rosli Sukri [EMAIL PROTECTED] wrote:
u need another box say box a with real/addressable ip address. create an iax entry in box a and have the private ip (box b) box register to box a. then you can do a Dial(IAX2/boxb/${EXTEN}) that will ring the extension connected to
your 192.168.100.249 boxhope that helps;)
On 11/12/06, nik600 [EMAIL PROTECTED]
wrote:
Hii have to forward a call from my asterisk server on another server but my server is behind nat.How can i setup my extension.conf?Actually i have set up it as follows:exten = 046566,1,Dial(
SIP/[EMAIL PROTECTED])my server has a private ip 192.168.100.249 and doesn't have a public ipIf i try to call
SIP/[EMAIL PROTECTED] from an adsl connection (with amodem, without nat) the call is routed succesfuly.If i try to forward the call from my server i cant route the call... (i send many INVITE but without any answer)
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[asterisk-users] Knowing when an answerphone answers
Hi all, I have found that when I use an announcement at the start of the call it results in a useless answerphone message if the call goes onto answerphone for any reason - the message being a chopped off version of the announcement. Does anyone know of a good way to detect that an answerphone has answered - or how to detect the tone that they typically use so I can code around this (with a different message). Thanks, Nic ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Outgoing problem on PRI
Dear All,
The resolution to the problem below was very easy and I guess that what
made it very hard:
callerid=asreceived
signalling=pri_cpe
switchtype=> euroisdn
context=from-zaptel
group=0
channel=>1-15,17-31
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear All,
I have an asterisk server version 1.2.12.1 along with trixbox
and I am having this nasty problem, I have a TE200P and have an E1 pri
attached to it and zttool says it's OK, I have configured the whole
31 channels into one group as follow:
Zapata-auto.conf:
callerid=asreceived
signalling=pri_cpe
switchtype=euroisdn
context=from-zaptel
group=0
channel=>1-15,17-31
/etc/zaptel.conf:
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
Now I can recieve calls on the pri and everyhting is well but I can't
make calls from the pri, whenever I try I get all circuits are busy message
here is a log from asterisk cli when I try to make a call out using pri
it is a tiny long but trixbox does add many macros and stuff put
I do have suspicions about what can cause the zap channel to get a Hungup
request as it seems from below that is the case :
-- Executing Macro("SIP/146-b78060b0", "dialout-trunk|3|6536595||")in
new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?3:2")
in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/146-b78060b0", "user-callerid")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?start")
in new stack
-- Executing Set("SIP/146-b78060b0", "REALCALLERIDNUM=146")
in newstack
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSER=146")
in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSERCIDNAME=Mohamed
Samir -UNIX") in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing Set("SIP/146-b78060b0", "CALLERID(all)=Mohamed
Samir - UNIX 146>") in new stack
-- Executing NoOp("SIP/146-b78060b0", "Using CallerID
"Mohamed Samir - UNIX" 146>") in new stack
-- Executing Macro("SIP/146-b78060b0", "record-enable|146|OUT")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0 > 0?2:4")
in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/146-b78060b0", "recordingcheck|20061110-162404|1163168644.20")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
-- Executing Macro("SIP/146-b78060b0", "dialout-trunk|3|6536595||")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?3:2")
in new stack
-- Goto (macro-dialout-trunk,s,3)
-- Executing Macro("SIP/146-b78060b0", "user-callerid")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?start")
in new stack
-- Executing Set("SIP/146-b78060b0", "REALCALLERIDNUM=146")
in new stack
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSER=146")
in new stack
-- Executing Set("SIP/146-b78060b0", "AMPUSERCIDNAME=Mohamed
Samir - UNIX") in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0?report")
in new stack
-- Executing Set("SIP/146-b78060b0", "CALLERID(all)=Mohamed
Samir - UNIX 146>") in new stack
-- Executing NoOp("SIP/146-b78060b0", "Using CallerID
"Mohamed Samir - UNIX" 146>") in new stack
-- Executing Macro("SIP/146-b78060b0", "record-enable|146|OUT")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "0 > 0?2:4")
in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/146-b78060b0", "recordingcheck|20061110-162404|1163168644.20")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheckrecordingcheck|20061110-162404|1163168644.20:
Outbound recording not enabled
recordingcheck|20061110-162404|1163168644.20: Outbound recording
not enabled
-- AGI Script recordingcheck completed, returning
0
-- AGI Script recordingcheck completed, returning
0
-- Executing NoOp("SIP/146-b78060b0", "No recording
needed") in new stack
-- Executing NoOp("SIP/146-b78060b0", "No recording
needed") in new stack
-- Executing Macro("SIP/146-b78060b0", "outbound-callerid|3")
in new stack
-- Executing Macro("SIP/146-b78060b0", "outbound-callerid|3")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?start")
in new stack
-- Executing GotoIf("SIP/146-b78060b0", "1?start")
in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Goto (macro-outbound-callerid,s,3)
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing NoOp("SIP/146-b78060b0", "REALCALLERIDNUM
is 146") in new stack
-- Executing Set("SIP/146-b78060b0", "USEROUTCID=")
in new stack
-- Executing Set("SIP/146-b78060b0", "USEROUTCID=")
in new stack
-- Executing Set("SIP/146-b78060b0", "EMERGENCYCID=")
in new stack
-- Executing Set("SIP/146-b78060b0", "EMERGENCYCID=")
in new stack
-- Executing Set("SIP/146-b78060b0", "TRUNKOUTCID=")
in new stack
-- Executing Set("SIP/146-b78060b0",
Re: [asterisk-users] sip forward behind a nat
On 11/12/06, Vicky [EMAIL PROTECTED] wrote: Yep make the server with dynamic ip register to server with static ip ( sip or iax both will do but in sip keep nat=yes while making extension ) the problem is that the server with dynamic ip can't register on the other server! This is the situation: Server with SIP application (public_address) | | - - - Internet | | Firewall (NAT) | | Server Asterisk (private ip:192.168.100.249/public ip:public_address_2) | Analogic Board | Telecom I want to make a call from Server Asterisk to the server with SIP Application. The SIP Application can't register to Server Asterisk (because the application can't do it, i know, it isn't a good thingbut this is the application) When The SIP Application receives a SIP call it responds (because a dummy SIP user is autoregistered on hisself) So i only have to make a call to SIP/[EMAIL PROTECTED] I've also tried to setup an asterisk server on my laptop, and make a call to SIP/[EMAIL PROTECTED] from the public_address network. It works! I only have to setup the Asterisk server in production to make a SIP call throw the NAT but without any SIP user registered on it. Can i do that? Many thanks to all ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Soundfiles adding during phone calls
Tom Lynn wrote: Ron, The guy is trying to help you. Tom, I believe it! Go to the link and read it. There is a feature that you can use to play a recording into the voice channel. Mine is set so when you press #9, the caller hears the lots of monkeys recording. I am not sure if that is correct: feature.conf: [applicationmap] shout2caller = *911,callee,Playback,shout-100dB ;Shout to caller if *911 was pressed - use 'callee' or 'caller' ask4name-Chinese = *910,callee,Playback,ask4name-Chinese; Ask caller for her/his name in Chinese and in extensions.conf and where should Set(DYNAMIC_FEATURES=hangup#play#testfeature) be and I want that only 601 and 621 can use this feature. bye Ronald Wiplinger The best part of it is that you can hang up and the recording will continue to play to the caller. When it expires, so does the call On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: http://www.voip-info.org/wiki-Asterisk+config+features.conf ... and where exactly did you see this feature bye Ronald Wiplinger On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I want to add some sound filed on demand during a phone call only possible on some extension numbers. I get many phone calls from local companies, but don't understand Chinese! I would like to record the call, but also ask the caller some questions, which should be added into the call with some keys on the phone, ... e.g. *66554 should add into the call: How are you? or What is your phone number? But I do have another application for that too. I get many fake phone calls, where Chinese people tell you that your phone bill is not paid, your court fee is not paid, and ask the caller to go to the ATM machine and key in a series of key strokes, most likely it will clear out your account. For such fake callers I would like to add a terrible noise to the call and make scare them as much as possible. Such fake calls I get now for each of my phone lines at least 10 each!!! Either the caller-id is not set, is 0 or is a tollfree number. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0647-0, 2006/11/09 Tested on: 2006/11/11 �U�� 11:07:21 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus:
Re: [asterisk-users] Asterisk architecture
je . a écrit :
Thanks for your response. I'm looking specifically at
Asterisk in a SIP-only implementation. So no need for
Asterisk to transfer calls between PSTN and SIP. Is
there in such a case still a need for a PBX/Asterisk?
As you can see in a typical SIP communication diagram, Asterisk (proxy,
register, ...) can be useless during the communication (communications
over RTP). But before having the communication, you must do different
things :
-- find your correspondant : if you know his IP adress, you can call
him directly, if no, you need something to find him for you, with
information you have (generaly his name). It is like DNS on the web, you
can connect you to a web server if you know its IP adress, but it is
less comfortable.
-- try to establish a communication to him : with IP adress, you can.
But if he is busy, or if he does not answer, the communication is over.
But maybe you want a telephony service (and not only voice service), I
mean forward to voicemail on busy, forward on voicemail if no answer,
tell to try 10 sec and then pass to an another phone number, if you want
the caller listen a message (Welcome on MegaWorld company, we will you
connect to the commercial service), if you want to make the call
ringing 5 phones in the same time (like call center), ... for all theses
services, you need something between the two (or more) terminals (User
Agent). There Asterisk has his place for building a _dialplan_.
-- build a callcenter : make queues, define phone groups, implement a
CTI, you need Asterisk and its scripting power.
-- have a interactive vocal server : you need Asterisk.
-- you want security : you should want to forbid person to call some
phones (if you want to call the director from your phone, you will
directly routed to the secretary. If the call come from the financial
director the the enterprise director, the call will not be rerouted).
-- ...
You must see Asterisk like an intermediate for adding _telephony_ services :
{phone1} {asterisk} -- {phone2}
{UAC}{UAS|add service|UAC} ---{UAS}
From what I understand, as long as two users know
eachother's IP addresses there's no need for Asterisk,
is this correct? I'm talking here about a pure SIP
implementation.
so if you have u1 and u2 and they simply wish to
communicate and they have eachother's IP, do they need
Asterisk for something?
Also, in a pure SIP implementation, what does Asterisk
offer and what is Asterisk required for?
Finaly, if you want make a simple call, no need to have Asterisk. If you
want to make simple calls with more comfort (call by name and not by
@IP), install Asterisk. If you want some _telephony_services_ you need
Asterisk.
Bye
GL
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Re: [asterisk-users] Knowing when an answerphone answers
Am Sonntag, den 12.11.2006, 08:50 + schrieb Nic Hughes: Hi all, I have found that when I use an announcement at the start of the call it results in a useless answerphone message if the call goes onto answerphone for any reason - the message being a chopped off version of the announcement. Does anyone know of a good way to detect that an answerphone has answered - or how to detect the tone that they typically use so I can code around this (with a different message). http://www.mail-archive.com/asterisk-users% 40lists.digium.com/msg161611.html http://www.voip-info.org/wiki/view/Asterisk+cmd+BackGroundDetect http://lists.digium.com/pipermail/asterisk-dev/2006-March/019086.html might be of interest. It seems you were not the first person with that idea. HTH Anselm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dynamically modifying the dialplan?
Hi Brian, many thanks to you for your answers in the past! The always gave me the little bit of mising information... My Asterisk box is running fine now so I want to try the next step... And now to all of you What I want to implement is to use 1 button of my snom-360 phone for following purpose: If I leave my desk I press this button. A light should show up as an indicator/reminder. From this moment all calls to my extension should immediately be transferred to my voicemail box. When I return I press the button again, the light goes off and all calls to my extension should ring my phone again. Now, can I achieve this with a static dialplan in extensions.conf or do I have to use all that Realtime + DB magic? Many thanks, Norbert ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dynamically modifying the dialplan?
I think its same as DND (do not disturb ) . It can be activated by *78 and deactivated by *79 . I use freepbx for configuration so i am not sure if its there in default asterisk setup . I snipped some part of my configuration from freepbx's config files
[app-dnd-on]exten = *78,1,Answerexten = *78,n,Wait(1)exten = *78,n,Macro(user-callerid,)exten = *78,n,Set(DB(DND/${CALLERID(number)})=YES)exten = *78,n,Playback(do-not-disturbactivated)
exten = *78,n,Macro(hangupcall,)[app-dnd-off]exten = *79,1,Answerexten = *79,n,Wait(1)exten = *79,n,Macro(user-callerid,)exten = *79,n,dbDel(DND/${CALLERID(number)})exten = *79,n,Playback(do-not-disturbde-activated)
exten = *79,n,Macro(hangupcall,)On 12/11/06, Norbert Zawodsky [EMAIL PROTECTED]
wrote:
Hi Brian,many thanks to you for your answers in the past! The always gave me thelittle bit of mising information...My Asterisk box is running fine now so I want to try the next step...And now to all of you
What I want to implement is to use 1 button of my snom-360 phone forfollowing purpose:If I leave my desk I press this button. A light should show up as anindicator/reminder. From this moment all calls to my extension should
immediately be transferred to my voicemail box.When I return I press the button again, the light goes off and all callsto my extension should ring my phone again.Now, can I achieve this with a static dialplan in
extensions.conf or doI have to use all that Realtime + DB magic?Many thanks,Norbert___--Bandwidth and Colocation provided by
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Re: [asterisk-users] dynamically modifying the dialplan?
On 19:28, Sun 12 Nov 06, Vicky wrote: I think its same as DND (do not disturb ) . It can be activated by *78 and deactivated by *79 . I use freepbx for configuration so i am not sure if its there in default asterisk setup . I snipped some part of my configuration from freepbx's config files Isn't DND implemented in the snom itself ? I know it is on the grandstream gxp2000 and all my cisco phones. Hard to believe the snom does not have this. Actually it is :) http://www.snom.com/download/man_snom360_en.pdf page 40 shows how it works. It will also put some info on the screen telling you the phone is in DND mode. -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Example Polycom function key config
Jamie Heckford wrote: HOWEVER It doesn't seem to send them to the *current* call. It places the current call on hold and tries to place a call on a new line. Currently looking for a workaround to this, will let you know. Jamie, Were you able to get a workaround on this? Just curious, Doug. -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [asterisk-users] Latest Debian and latest zaptel
Hi, The page can not be found! Many thanks, Christian On 2006-11-12 at 05:39 Tzafrir Cohen wrote: On Sat, Nov 11, 2006 at 09:42:27PM +0200, Tzafrir Cohen wrote: I wonder, though, how that symlink was created. I hope an advice by me was not involved... I guess that more than just advice. zaptel-source and libtonezoe-dev versions 1.2.9.1.dfsg-2, 1.2.10.dfsg-1 and 1.2.10.dfsg-2 contain the relevant symlinks. I've just removed those fixes (aimed originally at making asterisk 1.4 buildable with that package). Anyway, expect 1.4 debs soon. Or take a look at ones from http://www.principaltelecom.com.br/asterisk-1.4 -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1862 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk-addons 1.4 SVN fails to compile
It seems like asterisk-addons in SVN has been broken for the last few weeks: gcc -DHAVE_CONFIG_H -I. -I. -I. -I./ooh323c/src -I./ooh323c/src/h323 - DGNU -D_GNU_SOURCE -D_REENTRANT -D_COMPACT -c src/chan_h323.c -MT chan_h323.lo -MD -MP -MF .deps/chan_h323.TPlo -fPIC -DPIC -o .libs/ chan_h323.lo src/chan_h323.c: In function 'ooh323_new': src/chan_h323.c:250: error: too few arguments to function 'ast_channel_alloc' ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Speeding up SayDigits?
I would like SayDigits to say a phone number faster. Is there a way to control the speed? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Looking for a simple TFTP server for Linux
Hi, I am looking for a TFTP server that is easy like the tftpd32 for Windows that I have been using. Just want to start it with a command and my Cisco can connect and retreive the config files from it. Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a simple TFTP server for Linux
-Original Message- From: Christian [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Sun, 12 Nov 2006 16:27:42 +0100 Subject: [asterisk-users] Looking for a simple TFTP server for Linux Hi, I am looking for a TFTP server that is easy like the tftpd32 for Windows Most distros come with a tftp. What distro are you working with? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Media Gateway
I am planning to use asterisk with Digium TDM2404E card as a media gateway to terminate traffic to Cell phones. Anyone got this working before with no problmes, specially with Answer/Disconnect supervision? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [asterisk-users] Looking for a simple TFTP server for Linux
Hello Doug, I'm using Debian. Many thanks, Christian On 2006-11-12 at 15:34 Doug Lytle wrote: -Original Message- From: Christian [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Sun, 12 Nov 2006 16:27:42 +0100 Subject: [asterisk-users] Looking for a simple TFTP server for Linux Hi, I am looking for a TFTP server that is easy like the tftpd32 for Windows Most distros come with a tftp. What distro are you working with? Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ NOD32 1862 (20061110) Information __ This message was checked by NOD32 antivirus system. http://www.eset.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a simple TFTP server for Linux
On Sun, Nov 12, 2006 at 04:27:42PM +0100, Christian wrote: Hi, I am looking for a TFTP server that is easy like the tftpd32 for Windows that I have been using. Just want to start it with a command and my Cisco can connect and retreive the config files from it. Debian has: tftpd tftpd-hpa atftpd tftpd-hpa is H. Peter Anvin's take on tftpd. It is slightly larger, but has some handy features. atftpd has even more. A sore point to both tftpd and tftpd-hpa is the logging, which is a bit lacking. TFTP is a simple protocol, and by default there's no configuration: # installing both client and server, to be able to test server with # client: apt-get install tftpd-hpa tftp-hpa # Should be working now. Let's test: echo hi /var/lib/tftpboot/hello tftp localhost -c get /hello # Should complain if there is a problem. Did we get the file? cat hello # we get hi -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: dynamically modifying the dialplan?
NZ == Norbert Zawodsky [EMAIL PROTECTED] writes:
NZ If I leave my desk I press this button. A light should show up as
NZ an indicator/reminder. From this moment all calls to my extension
NZ should immediately be transferred to my voicemail box.
NZ When I return I press the button again, the light goes off and all
NZ calls to my extension should ring my phone again.
Apart from the light it's very easy to achieve.
12345,1,Set(DB(desk/away)=${IF($[${DB(desk/away)}=away]?back:away)}
12345,n,Hangup
(Did I ever mention that I hate dialplan syntax? There's probably a
dozen bracket or quotation errors in the above, and it took 10 minutes
to write just that one line.)
Check in the dial plan when you try to dial the phone whether
DB(desk/away) is away or back, and go to voicemail. Then set your
phone up to dial 12345 when that particular button is pressed.
An alternative is to make an extension which goes to voice mail
directly, and simply redirect the phone to that extension. It's a bit
more than one button, but at least the Snom 360 will show that the
redirection is active. Perhaps the magical function buttons can even
be programmed to do it.
/Benny
___
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[asterisk-users] Re: dynamically modifying the dialplan?
BA == Benny Amorsen [EMAIL PROTECTED] writes: BA An alternative is to make an extension which goes to voice mail BA directly, and simply redirect the phone to that extension. It's a BA bit more than one button, but at least the Snom 360 will show that BA the redirection is active. Perhaps the magical function buttons BA can even be programmed to do it. Or I could wake up and realize that DND is exactly what is wanted here. Doh! /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Some pictures from Astricon 2006 in Dallas
Some pictures from Astricon 2006 in Dallas.http://gallery.lith.za.net/Astricon-2006-- RegardsRob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CLI message: remote unix connection disconnected
snip I am running the most recent asterisk 1.2.13 on a Fedora 3.0. When I go into asterisk (asterisk -r), defaults to verbose 3 and I get a stream of messages: Remote Unix connection Remote Unix connection disconnected ... ... -- end of file --- has anybody faced this problem? is there a method to get around it? what else could be causing it? ___ /snip This comes from either using the asterisk manager (manager.conf) or when you asterisk -rx ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Some pictures from Astricon 2006 in Dallas
Hey Rob, thanks for that. Brought back good memories. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Lith Sent: Sunday, 12 November 2006 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Some pictures from Astricon 2006 in Dallas Some pictures from Astricon 2006 in Dallas. http://gallery.lith.za.net/Astricon-2006 -- Regards Rob ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VM problems...
Hi, perhaps someone has an answer for me: - voicemail isn't sending any mails, even SMTP_SERVER and SMTP_Domain are configured in rc.conf, and the mail address ist configured in voicemail.conf. - On my grandstream GPX2000 VM light flashes when there is a new voicemail, but it doesn't go out after I have deleted the message best regards -- Alexander Topolanek Intelligente EDV- und Telekommunikationslösungen Montage von Sicherheitstechnik Probst Peitlstr. 85 2103 Langenzersdorf +43 664 507 68 09 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Looking for a simple TFTP server for Linux
Most Linux distros have a TFTP server built in, however usually it functions through xinetd, which is probably already running on your machine, so actually it would cause no extra usage on your system unless the TFTP was in use. Check your distro's package repository and you should find something like tftpd On 11/12/06, Christian [EMAIL PROTECTED] wrote: Hi,I am looking for a TFTP server that is easy like the tftpd32 for Windows that I have been using. Just want to start it with a command and my Cisco can connect and retreive the config files from it.Many thanks, Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: Choppy sound in voicemail using Asterisk1.2.11 on CENTOS4 guest on vmware server
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Martin Joseph Sent: Saturday, November 11, 2006 15:01 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Choppy sound in voicemail using Asterisk1.2.11 on CENTOS4 guest on vmware server On 2006-11-10 09:10:30 -0800, Mario François Jauvin [EMAIL PROTECTED] said: This is a multi-part message in MIME format. I have had no success in getting the voicemail working on Asterisk 1.2.11 on CENTOS4(2.6 kernel) guest on vmware server 1.0.1. I tried with or without ztdummy device, renice -20 on asterisk process and even real-time priority on the host Windows XP box for the vmware process. I am running on an AMD Athlon 64 X2 4600+. The behaviour is when the voicemail answer, the voice sound ok but when asterisk starts saying the digits from the extension, the sound starts becoming very choppy. The voice after the digits is still choppy. Does anyone have a suggestion? The codec that asterisk is using with the softphone I am using is the GSM codec. If it sounds like the trouble starts when digits are being spoken, perhaps it's related to your disk controller? Is there a zap channel involved in this setup (ie how is the phone connected)? Don't know much about the whole vmware thing, but it sounds like an interrupt issue perhaps? Marty You don't mention what version of kernel you are using. There are known issues with certain versions of Kernel when running inside VMWARE. 2.6.9-34 branch works well, 2.6.9-42 does not! Andrew ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outgoing works, incoming fails on asterisk passthrough to inter-tel
Hi everybody,
Well, I've finally got asterisk to to talk nicely with my Intertel pbx.
Currently there is a outside T1 line (em wink start, esf, b8zs)
connected to asterisk, and then asterisk connected similarly to my
Intertel pbx. For right now all asterisk is doing is passing calls
between the two.
When I call out from the pbx, I can connect perfectly to the outside
world. When I call from outside, I can talk to the asterisk box, but
asterisk fails to pass the call to the pbx. The following is the log of
the connection (numbers scrambled to protect the innocent). At the end
I've included my extensions.conf file. The incoming phone number is
801-555-, and I'm calling 555-5154
I've tried changing the exten = _X.,2,Dial(Zap/g3/${EXTEN},15,r)
line that transfers the call to the pbx to exlude the , add in the
caller id, and various other things, but the results are always
identical. If anyone has any experience with talking to inter-tel pbx's,
please let me know what trick is necessary.
Thanks a million.
call log follows:
Nov 12 12:48:35 VERBOSE[32609] logger.c: -- Starting simple switch
on 'Zap/73-1'
Nov 12 12:48:36 DEBUG[32609] chan_zap.c: DTMF digit: * on Zap/73-1
Nov 12 12:48:36 DEBUG[32609] chan_zap.c: DTMF digit: 8 on Zap/73-1
Nov 12 12:48:36 DEBUG[32609] chan_zap.c: DTMF digit: 0 on Zap/73-1
Nov 12 12:48:36 DEBUG[32609] chan_zap.c: DTMF digit: 1 on Zap/73-1
Nov 12 12:48:36 DEBUG[32609] chan_zap.c: DTMF digit: 5 on Zap/73-1
Nov 12 12:48:36 DEBUG[32609] chan_zap.c: DTMF digit: 5 on Zap/73-1
Nov 12 12:48:36 DEBUG[32609] chan_zap.c: DTMF digit: 5 on Zap/73-1
Nov 12 12:48:37 DEBUG[32609] chan_zap.c: DTMF digit: 5 on Zap/73-1
Nov 12 12:48:37 DEBUG[32609] chan_zap.c: DTMF digit: 5 on Zap/73-1
Nov 12 12:48:37 DEBUG[32609] chan_zap.c: DTMF digit: 5 on Zap/73-1
Nov 12 12:48:37 DEBUG[32609] chan_zap.c: DTMF digit: 5 on Zap/73-1
Nov 12 12:48:37 DEBUG[32609] chan_zap.c: DTMF digit: * on Zap/73-1
Nov 12 12:48:37 DEBUG[32609] chan_zap.c: DTMF digit: 1 on Zap/73-1
Nov 12 12:48:37 DEBUG[32609] chan_zap.c: DTMF digit: 5 on Zap/73-1
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: DTMF digit: 4 on Zap/73-1
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: DTMF digit: * on Zap/73-1
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: Enabled echo cancellation on
channel 73
Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Executing
Goto(Zap/73-1, to-intertel|154|1) in new stack
Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Goto (to-intertel,154,1)
Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Executing
SetTransferCapability(Zap/73-1, SPEECH) in new stack
Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Setting transfer
capability to: 0x00 - SPEECH.
Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Executing
Dial(Zap/73-1, Zap/g3/154|15|r) in new stack
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: Dialing '154'
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: Deferring dialing...
Nov 12 12:48:38 VERBOSE[32609] logger.c: -- Called g3/154
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: Requested indication 3 on
channel Zap/73-1
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: Exception on 65, channel 49
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: Got event Wink/Flash(3) on
channel 49 (index 0)
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: Ignoring wink on channel 49
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: Exception on 65, channel 49
Nov 12 12:48:38 DEBUG[32609] chan_zap.c: Got event Hook Transition
Complete(12) on channel 49 (index 0)
Nov 12 12:48:40 DEBUG[32609] chan_zap.c: Exception on 65, channel 49
Nov 12 12:48:40 DEBUG[32609] chan_zap.c: Got event Dial Complete(9) on
channel 49 (index 0)
Nov 12 12:48:40 DEBUG[32609] chan_zap.c: Enabled echo cancellation on
channel 49
Nov 12 12:48:40 DEBUG[32609] chan_zap.c: Engaged echo training on channel 49
Nov 12 12:48:41 DEBUG[32609] chan_zap.c: Exception on 65, channel 49
Nov 12 12:48:41 DEBUG[32609] chan_zap.c: Got event Dial Complete(9) on
channel 49 (index 0)
Nov 12 12:48:41 DEBUG[32609] chan_zap.c: Echo cancellation already on
Nov 12 12:48:54 VERBOSE[32609] logger.c: -- Nobody picked up in 15000 ms
Nov 12 12:48:54 DEBUG[32609] chan_zap.c: Requested indication -1 on
channel Zap/73-1
Nov 12 12:48:54 DEBUG[32609] chan_zap.c: Hangup: channel: 49 index = 0,
normal = 65, callwait = -1, thirdcall = -1
Nov 12 12:48:54 DEBUG[32609] chan_zap.c: disabled echo cancellation on
channel 49
Nov 12 12:48:54 DEBUG[32609] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/49-1
Nov 12 12:48:54 DEBUG[32609] chan_zap.c: Updated conferencing on 49,
with 0 conference users
Nov 12 12:48:54 VERBOSE[32609] logger.c: -- Hungup 'Zap/49-1'
Nov 12 12:48:54 DEBUG[32609] app_dial.c: Exiting with DIALSTATUS=NOANSWER.
Nov 12 12:48:54 VERBOSE[32609] logger.c: -- Executing
Playback(Zap/73-1, vm-nobodyavail) in new stack
Nov 12 12:48:54 DEBUG[32609] chan_zap.c: Took Zap/73-1 off hook
Nov 12 12:48:54 DEBUG[32558] channel.c: Avoiding initial deadlock for
'Zap/73-1'
Nov 12 12:48:54
[asterisk-users] Determine if Call is from a cellular phone
Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk billing
I am having asterisk working with cdr mysql patch and freepbx for configurations . It stores all records in mysql tables and i can do further post paid billing myself . I am looking for a simple system that can show a user live call logs via web interfaceasperaccountcodeonsipextensions(muchlikeasterisk-stats)butitshouldnotshowalllogsatsametime..ineeditasperaccountcode(likeextension777-999hasaccountcodeaccount1soitwouldshowonlycallsofthisextensionfrommysqltable( on some php page ). Ihaveseenastppandotherbillingsystemsbutidontneedthatmuchfunctionalityorcomplexityrightnow.Can someone suggest the easiest way to do this ?? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determine if Call is from a cellular phone
What exactly are you trying to do? YOu can determine where the number was assigned and if it was originally a cell phone easily and cheapOn 11/12/06, Dovid B [EMAIL PROTECTED] wrote: Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determine if Call is from a cellular phone
With LNP in the US, there really is no way to determine if the call originates from a landline, a VOIP line or a wireless line Most numbers are portable and even the NPA doesn't tell much any more. I have a wireless phone with a ( former ) landline number, and a VOIP line with a ( former ) wireless number. I have to wonder why you care. John Novack Dovid B wrote: Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] same extension on softphones and hardphones
Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the extension is dialed. The sip devices they will be using are Grandstream GXP2000 desktop phones and Xten Eyebeam softphones. Each user will have one of each. What is the best way to accomplish this? Xten eyebeam ext 110 \ \ -- Asterisk 1.2.8 / GXP2000 phone ext 110 / Need both phones to ring when extension 110 is dialed. Is this possible without creating ring groups? Thanks in advance. JR ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Random 'no audio' problem
Matt, as a start, what I can advise you is to take a tethereal trace and try to reproduce the problem. nohup tethereal host a.b.c.d -s2000 -w /tmp/yourtrace.cap Where a.b.c.d is the IP address of your IP phone. You can then analyse the trace and at least see if the asterisk box is sending AND receiving RTP traffic to and from the phone. We have seen some issues in the past with 'no audio' or 'unidirectional audio' due to wrong firmware versions in SIP phones or due to ethernet switch instability, even on a cisco 3560 switches. Hope this helps, Jordi Matt wrote: I have no idea.. that sounds like your Internet connection is going down and leaving you for a bit and then coming back. My issue is a local network connection, no public Internet... or you can even call in from outside on the PSTN and the audio, both ways, will just stop. On 11/3/06, Zeeshan Zakaria [EMAIL PROTECTED] wrote: I have the same problem with IAX trunk and SIP extensions. Now I think its the IAX. I never had this problem om SIP trunk. Am I right? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jordi Nelissen E S C A U X Business IP Telephony www.escaux.com -- Email from people at escaux.com does not usually represent official policy of ESCAUX. See http://www.escaux.com/disclaimer for details. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] operator console
Check out the ESCAUX net.PBX operator console. In use in various companies with 200+ extensions. Powerfull and convenient. http://www.escaux.com//index.php?option=com_contenttask=viewid=61Itemid=350 Best Regards, Jordi -- www.escaux.com Business IP Telephony Forrest Beck wrote: Talk to the folks at Asteria. The have a product called Reign. It looks just like your old interface, runs off .NET as a client on the machine. http://www.asteriasgi.com/pbx/reign On 11/7/06, Stephen Wingfield [EMAIL PROTECTED] wrote: Andres, The Bicom Systems Operator Panel is probably what you are looking for. OPCOM http://www.bicomsystems.com/docs/opcom/1.0/html/ This is included with every copy of PBXware and is fully supported. If you care to register you may order a trial of PBXware with our SOHO. Regards Steve steve 'at' bicomsystems 'dot' com - Original Message - From: Andres Paglayan To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, October 30, 2006 5:27 PM Subject: [asterisk-users] operator console Hi, My users are currently using an operator console interface like this: see it at: http://www.whssf.org/interface.jpg which came with a Praxon PDX we got about 5 years ago, which is now unsupported, it works very good and converts any analog phone plugged into the system into a powerful console, (provided you have a computer next to it) you just provide the box ip, user login, user pass, and extension, and voila. I'll be switching the company's phone system to Asterisk. I know * is way much more flexible and rich featured than the box we currently have, ...but I'll need to give the users a good mean to see what's going on, who is busy, easy transfer with click here and there, easy conference, easy queue handler, easy way to see/use many lines at the same time is there any best console they can use? I don't mind using a commercial product, if the only part we have to pay for is the gui, besides, we will buying the enterprise * version Thanks a bunch, Andres ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jordi Nelissen E S C A U X Business IP Telephony www.escaux.com -- Email from people at escaux.com does not usually represent official policy of ESCAUX. See http://www.escaux.com/disclaimer for details. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Determine if Call is from a cellular phone
Depending on how you connect to the PSTN and what type of call is being made, you may have access to the ANI II digits. The II digits tell you what type user/service originated the call from such as: regular phone, hotel/motel guest phone, pay phone, inmate phone, and various types of cellular/PCS phones. See http://www.nanpa.com/number_resource_info/ani_ii_assignments.html for the digits values and what they mean. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Sunday, November 12, 2006 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Determine if Call is from a cellular phone With LNP in the US, there really is no way to determine if the call originates from a landline, a VOIP line or a wireless line Most numbers are portable and even the NPA doesn't tell much any more. I have a wireless phone with a ( former ) landline number, and a VOIP line with a ( former ) wireless number. I have to wonder why you care. John Novack Dovid B wrote: Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determine if Call is from a cellular phone
It's for a call center. Calls are routed based on location. The customer would rather the to be transferd without human interaction unless abolutely nesc. - Original Message - From: John Novack [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, November 12, 2006 11:26 PM Subject: Re: [asterisk-users] Determine if Call is from a cellular phone With LNP in the US, there really is no way to determine if the call originates from a landline, a VOIP line or a wireless line Most numbers are portable and even the NPA doesn't tell much any more. I have a wireless phone with a ( former ) landline number, and a VOIP line with a ( former ) wireless number. I have to wonder why you care. John Novack Dovid B wrote: Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] same extension on softphones and hardphones
Am Sonntag, den 12.11.2006, 16:29 -0500 schrieb Jeronimo Romero:
Sorry if you see this message repeated twice. I would like to set up
hard phones and softphones with the same extension so that when anybody
in the company dials an extension, each user's hardphone and softphone
will ring at the same time. I've tried setting this up before, but I
noticed that the last sip device to register with the same extension is
the only one that rings when the extension is dialed. The sip devices
they will be using are Grandstream GXP2000 desktop phones and Xten
Eyebeam softphones. Each user will have one of each. What is the best
way to accomplish this?
Xten eyebeam ext 110 \
\
-- Asterisk 1.2.8
/
GXP2000 phone ext 110 /
One possible solution is to have one sip account for each _device_, not
extension; say sip110h and sip110s for the 110-user.
Then use the dial command in your extensions.conf like
exten = _1XX,1,Dial(SIP/sip${EXTEN}sSIP/sip${EXTEN}h)
This will cause parallel ringing phones. First come first serve.
Hth
Anselm
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Re: [asterisk-users] best gui
Just give ESCAUX net.PBX Free Edition a try. You can start checking out our GUI at http://smp.free.escaux.com. These web interfaces will generate the asterisk config files that are then pushed to your asterisk box. The full solution can be downloaded at http://www.escaux.com/netpbx Have fun, Jordi [EMAIL PROTECTED] wrote: On Sat, 4 Nov 2006 06:36:06 -0500 Zeeshan Zakaria [EMAIL PROTECTED] wrote: Trixbox www.trixbox.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jordi Nelissen E S C A U X Business IP Telephony www.escaux.com -- Email from people at escaux.com does not usually represent official policy of ESCAUX. See http://www.escaux.com/disclaimer for details. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Determine if Call is from a cellular phone
Where can I get this info ? - Original Message - From: Andrew Joakimsen To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, November 12, 2006 11:18 PM Subject: Re: [asterisk-users] Determine if Call is from a cellular phone What exactly are you trying to do? YOu can determine where the number was assigned and if it was originally a cell phone easily and cheap On 11/12/06, Dovid B [EMAIL PROTECTED] wrote: Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Question about Mitel phones
Yes, the Mitel phones do have a Web interface for configuration. They also support mass-deployment scenarios with TFTP HTTP. You may want to check out these: http://sipdnld.mitel.com/ http://edocs.mitel.com/DB/5212_5224/WebConfigHelp_Admin_en_CA/WebHelp/ WebConfig.htm Thanks, - Jesse On Nov 10, 2006, at 10:35 AM, [EMAIL PROTECTED] wrote: Message: 9 Date: Fri, 10 Nov 2006 17:03:29 +0100 From: Christian [EMAIL PROTECTED] Subject: [asterisk-users] Question about Mitel phones To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 Hi all, Does anyone know if the Mitel phone features a webintreface for configuring the phone? Many thanks, Christian -- Jesse Peterson [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cadences zapata.conf
I edited zapata.conf to use custom ring cadences. Seemed to work, but upon some restarts, seems zapata.conf is not being read properly on startup as zap show cadences will show the defaults. Some restarts show the custom cadences. What's up with that? joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 one way audio
Experiencing one way audio using IAX2. I did see some other posts on this, and see there may be some internal issues with asterisk and one way audio. Can this be a widespread problem? So many seem to be using IAX, I find it puzzling. Some information points to this being a problem on asymmetrical connections. This is a decidedly asymmetrical connection, with 1.5 Mbs download and 256 kbs, upload. A satellite link, to boot. So, maybe this is a meltdown right from the start? Event the vendor of the IAX service was not too keen. Oddly, my first few connections worked fine (unexpectedly good audio, both ways). Being all happy and stuff, made a call to a client, to show off. Yep. could not hear me. Since then all calls have connected quickly, but are receive only. I've tried rebooting the asterisk box, changing jitter related stuff, no joy. It is behind a firewall, but I can see no packets dropped, related to the IP's involved. Anyway, if there are experiences to relate, please do. joe a. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] same extension on softphones and hardphones
Is this inherently an issue with sip? Is it possible for a sip server to
actually ring two different sip registration from the same account or is
this not possible under any sip enabled pbx?
Thanks again
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm
Martin Hoffmeister
Sent: Sunday, November 12, 2006 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] same extension on softphones and
hardphones
Am Sonntag, den 12.11.2006, 16:29 -0500 schrieb Jeronimo Romero:
Sorry if you see this message repeated twice. I would like to set up
hard phones and softphones with the same extension so that when
anybody
in the company dials an extension, each user's hardphone and softphone
will ring at the same time. I've tried setting this up before, but I
noticed that the last sip device to register with the same extension
is
the only one that rings when the extension is dialed. The sip devices
they will be using are Grandstream GXP2000 desktop phones and Xten
Eyebeam softphones. Each user will have one of each. What is the best
way to accomplish this?
Xten eyebeam ext 110 \
\
-- Asterisk 1.2.8
/
GXP2000 phone ext 110 /
One possible solution is to have one sip account for each _device_, not
extension; say sip110h and sip110s for the 110-user.
Then use the dial command in your extensions.conf like
exten = _1XX,1,Dial(SIP/sip${EXTEN}sSIP/sip${EXTEN}h)
This will cause parallel ringing phones. First come first serve.
Hth
Anselm
___
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RE: [asterisk-users] same extension on softphones and hardphones
This is not a SIP issue, but a problem with your configuration.
We have all hard phones register/authenticate with their MAC address as the
user/peer name. Soft phones use user id's that correspond to the person. We
then have our dialplan ring the appropriate devices (soft or hard) depending
on which extension was dialed.
Use the operator in the dial string to ring multiple devices.
Cheers,
Cullin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeronimo
Romero
Sent: Sunday, November 12, 2006 6:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] same extension on softphones and hardphones
Is this inherently an issue with sip? Is it possible for a sip server to
actually ring two different sip registration from the same account or is
this not possible under any sip enabled pbx?
Thanks again
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anselm Martin
Hoffmeister
Sent: Sunday, November 12, 2006 4:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] same extension on softphones and hardphones
Am Sonntag, den 12.11.2006, 16:29 -0500 schrieb Jeronimo Romero:
Sorry if you see this message repeated twice. I would like to set up
hard phones and softphones with the same extension so that when
anybody
in the company dials an extension, each user's hardphone and softphone
will ring at the same time. I've tried setting this up before, but I
noticed that the last sip device to register with the same extension
is
the only one that rings when the extension is dialed. The sip devices
they will be using are Grandstream GXP2000 desktop phones and Xten
Eyebeam softphones. Each user will have one of each. What is the best
way to accomplish this?
Xten eyebeam ext 110 \
\
-- Asterisk 1.2.8
/
GXP2000 phone ext 110 /
One possible solution is to have one sip account for each _device_, not
extension; say sip110h and sip110s for the 110-user.
Then use the dial command in your extensions.conf like
exten = _1XX,1,Dial(SIP/sip${EXTEN}sSIP/sip${EXTEN}h)
This will cause parallel ringing phones. First come first serve.
Hth
Anselm
___
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[asterisk-users] Headaches with Video over SIP
Greetings all, I'm playing with asterisk and two Polycom VSX300 videoconferencing units. And I'm having zero luck getting video working over SIP. The two units register fine with asterisk, and with allow=all in sip.conf, the two units establish voice. But no video. And no obvious messages as to whats going wrong. The config for each is (they're numbered 201 and 202): [202] secret= type=friend context=from-sip-202 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=all If you're wondering why I do the disallow=all immediately followed by allow=all, it's because the allow line has spent a lot of time with restricted codecs to see if that makes a difference. I can provide the full sip.conf, extensions.conf, and debug output if anyone wants to see them. Any suggestions as to where things are falling down? -- Peter Howard URSYS 13 Burwood Rd, Burwood, NSW 2134 Ph: 02 8745 2816Fax: 02 8745 2828 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headaches with Video over SIP
Oops, Asterisk version is 1.2.12 (on Ubuntu) On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote: Greetings all, I'm playing with asterisk and two Polycom VSX300 videoconferencing units. And I'm having zero luck getting video working over SIP. The two units register fine with asterisk, and with allow=all in sip.conf, the two units establish voice. But no video. And no obvious messages as to whats going wrong. The config for each is (they're numbered 201 and 202): [202] secret= type=friend context=from-sip-202 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=all If you're wondering why I do the disallow=all immediately followed by allow=all, it's because the allow line has spent a lot of time with restricted codecs to see if that makes a difference. I can provide the full sip.conf, extensions.conf, and debug output if anyone wants to see them. Any suggestions as to where things are falling down? -- Peter Howard URSYS 13 Burwood Rd, Burwood, NSW 2134 Ph: 02 8745 2816Fax: 02 8745 2828 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Determine if Call is from a cellular phone
I think puck.nether.net may still have a txt file with the CO broken down by NPA-NXX. You can then look at the carrier and know if it is Cell/LandLine. You can also X-ref the CO-list and get Lat/Long and or simply the zipcode to help you locate the caller. Not perfect but unless the majority of your customer;s customers are roaming it should be accurate to a metro area. Alex From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid B Sent: Sunday, November 12, 2006 5:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Determine if Call is from a cellular phone Where can I get this info ? - Original Message - From: Andrew Joakimsen To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Sunday, November 12, 2006 11:18 PM Subject: Re: [asterisk-users] Determine if Call is from a cellular phone What exactly are you trying to do? YOu can determine where the number was assigned and if it was originally a cell phone easily and cheap On 11/12/06, Dovid B [EMAIL PROTECTED] wrote: Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headaches with Video over SIP
On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote: Greetings all, I'm playing with asterisk and two Polycom VSX300 videoconferencing units. And I'm having zero luck getting video working over SIP. The two units register fine with asterisk, and with allow=all in sip.conf, the two units establish voice. But no video. And no obvious messages as to whats going wrong. The config for each is (they're numbered 201 and 202): [202] secret= type=friend context=from-sip-202 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=all If you're wondering why I do the disallow=all immediately followed by allow=all, it's because the allow line has spent a lot of time with restricted codecs to see if that makes a difference. I can provide the full sip.conf, extensions.conf, and debug output if anyone wants to see them. Any suggestions as to where things are falling down? Do you have videosupport=yes in your sip.conf? Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headaches with Video over SIP
On Mon, 2006-11-13 at 00:57 +0100, Patrick wrote: On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote: Greetings all, I'm playing with asterisk and two Polycom VSX300 videoconferencing units. And I'm having zero luck getting video working over SIP. The two units register fine with asterisk, and with allow=all in sip.conf, the two units establish voice. But no video. And no obvious messages as to whats going wrong. The config for each is (they're numbered 201 and 202): [202] secret= type=friend context=from-sip-202 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=all If you're wondering why I do the disallow=all immediately followed by allow=all, it's because the allow line has spent a lot of time with restricted codecs to see if that makes a difference. I can provide the full sip.conf, extensions.conf, and debug output if anyone wants to see them. Any suggestions as to where things are falling down? Do you have videosupport=yes in your sip.conf? Yes I do. I've also confirmed that I have a version of asterisk which includes the patch for H263P (which is what the Polycoms want to talk). -- Peter Howard URSYS 13 Burwood Rd, Burwood, NSW 2134 Ph: 02 8745 2816Fax: 02 8745 2828 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel compile problems
I'm having difficulties getting zaptel to compile. I've compiled it in the past and never had any difficulties to speak of, but on this particular machine I have problems. The OS configuration is the same as I've used in the past and the hardware is identical. Obviously there's some subtle difference (probably with the packages installed, but I haven't been able to locate anything). Everything works fine until I get to the xpp stuff: CC [M] /usr/src/zaptel-1.2.6/ztdummy.o CC [M] /usr/src/zaptel-1.2.6/xpp/card_fxs.o /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_card_new': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:98: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_card_init': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:114: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c:117: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_card_remove': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:134: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c:137: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_CHAN_ENABLE_send': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:190: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_CHAN_POWER_send': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:213: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_CHAN_CID_send': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:241: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_RING_send': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:264: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_SETHOOK_send': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:276: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_LED_send': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:318: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c:330: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_RELAY_OUT_send': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:350: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c:361: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_SLIC_INIT_send': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:377: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_SLIC_QUERY_send': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:403: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_SIG_CHANGED_handler': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:421: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c:426: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c:432: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_SLIC_REPLY_handler': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:456: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c:463: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `fxs_packet_is_valid': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:522: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `fxs_packet_dump': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:529: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `parse_slic_cmd': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:595: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c:607: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c:612: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `process_slic_cmdline': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:643: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `card_fxs_startup': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:687: error: invalid use of undefined type `struct module' /usr/src/zaptel-1.2.6/xpp/card_fxs.c:687: error: invalid use of undefined type `struct module' make[3]: *** [/usr/src/zaptel-1.2.6/xpp/card_fxs.o] Error 1 make[2]: *** [/usr/src/zaptel-1.2.6/xpp] Error 2 make[1]: *** [_module_/usr/src/zaptel-1.2.6] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.17.11-fai-p4' make: *** [linux26] Error 2 I've got all the documented prerequisites installed: bison bison-dev newt newt-dev Linux headers Linux kernel
[asterisk-users] Slow to get dialtone when going off hook - big problem for me :(
Hi All... My Asterisk system uses VoIP and also 2 POTS lines from Cox Communications. Recently, the dial tone presentation from Cox seems to have slowed, so it can take as long as 3 seconds to get a dial tone. The problem I am having is that Asterisk does not seem to wait for the dial tone when dialing out. I'm using zaptel T400 cards. Is there any way to configure it such that I can insert a delay between the time the card goes off hook and the time it starts dialing? Alternatively, can I make it wait until there is a dial tone? Incoming calls are just fine, so I am almost certain this is what's happening. Thanks! Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow to get dialtone when going off hook - big problem for me :(
Jim Archer wrote: Hi All... My Asterisk system uses VoIP and also 2 POTS lines from Cox Communications. Recently, the dial tone presentation from Cox seems to have slowed, so it can take as long as 3 seconds to get a dial tone. The problem I am having is that Asterisk does not seem to wait for the dial tone when dialing out. I'm using zaptel T400 cards. Is there any way to configure it such that I can insert a delay between the time the card goes off hook and the time it starts dialing? Alternatively, can I make it wait until there is a dial tone? Incoming calls are just fine, so I am almost certain this is what's happening. Thanks! Jim ___ add a couple or few w's before you dial. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow to get dialtone when going off hook - big problem for me :(
--On Sunday, November 12, 2006 10:06 PM -0500 Steve Totaro [EMAIL PROTECTED] wrote: add a couple or few w's before you dial. Okay, but where? I didn't see a w option for the dial command, and if I add a wait before the dial won;t that just delay going off hook? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow to get dialtone when going off hook -big problem for me :(
- Original Message -
From: Jim Archer [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, November 13, 2006 5:11 AM
Subject: Re: [asterisk-users] Slow to get dialtone when going off hook -big
problem for me :(
--On Sunday, November 12, 2006 10:06 PM -0500 Steve Totaro
[EMAIL PROTECTED] wrote:
add a couple or few w's before you dial.
Okay, but where? I didn't see a w option for the dial command, and if I
add a wait before the dial won;t that just delay going off hook?
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exten = X.,1,Dial(ZAP/1/www${EXTEN})
The w tells it to sait b4 sending the digits
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[asterisk-users] Trixbox dialout problems
Hello All.
I am trying to use RAGI the ruby agi framework with trixbox. I am
having a problem
with the dialout part. The RAGI framework creates a file in the
/var/spool/asterisk/outgoing directory and routes the call to an
extension (I have listed the relevent portion of the file below). The
problem is that the initial dial command does not execute properly in
trixbox. I am hoping somebody who has expertise in trixbox could help
me debug this problem. I can post an asterisk log snippet if anybody
is interested.
Here is the extension. The callout file is below too.
Extension_custom.conf
[dialout]
exten = outbound,1,Answer ; switches to outbound-handler
exten = outbound,2,Wait(60)
exten = outbound,3,Hangup
exten =
outbound-handler,1,Dial(${CallInitiate_phonenumber},50,gM(outbound-connect^${AGI_SERVER}${AGI_URL}^${CallInitiate_hashdata}^${MACHINE_STATUS_UNKNOWN}))
exten = outbound-handler,2,GotoIf($[${DIALSTATUS} = ANSWER]?104)
exten = outbound-handler,3,NoOp(status=${DIALSTATUS},
DIALEDTIME=${DIALEDTIME}, ANSWEREDTIME=${ANSWEREDTIME})
exten =
outbound-handler,4,SetVar(CallInitiate_hashdata=${CallInitiate_hashdata})
exten = outbound-handler,5,deadagi(agi://${AGI_SERVER}${AGI_URL})
;DIAL_STATUS is busy, etc.
exten = outbound-handler,6,Goto(104)
exten =
outbound-handler,102,SetVar(CallInitiate_hashdata=${CallInitiate_hashdata})
exten = outbound-handler,103,deadagi(agi://${AGI_SERVER}${AGI_URL})
;DIAL_STATUS is busy, etc.
exten = outbound-handler,104,Hangup()
[macro-outbound-connect]
exten = s,1,Answer()
exten = s,2,SetVar(CallInitiate_hashdata=${ARG2})
exten = s,3,SetVar(machinestatus=${ARG3})
exten = s,4,deadagi(agi://${ARG1})
exten = s,5,Hangup
Callout file-
;This file was generated by RAGI's callInitiate class
;File generated date: 11-07-2006 at 12:47 -- Tuesday
;Call date: 11-07-2006 at 12:47 -- Tuesday
Channel: Local/[EMAIL PROTECTED]
Callerid: 10
MaxRetries: 0
RetryTime: 5
WaitTime: 45
;magic extension for outbound calls via RAGI callInitiate
Context: dialout
Extension: outbound-handler
Priority: 1
SetVar: CallInitiate_phonenumber=90275524911
SetVar: CallInitiate_callerid=1000
SetVar: AGI_URL=/test_outbound/dialup
SetVar: AGI_SERVER=harborreach.panztel.biz:4573
SetVar: CallInitiate_hashdata=---+%0A
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Re: [asterisk-users] Slow to get dialtone when going off hook - big problem for me :(
Jim Archer wrote: Hi All... My Asterisk system uses VoIP and also 2 POTS lines from Cox Communications. Recently, the dial tone presentation from Cox seems to have slowed, so it can take as long as 3 seconds to get a dial tone. The problem I am having is that Asterisk does not seem to wait for the dial tone when dialing out. I'm using zaptel T400 cards. Is there any way to configure it such that I can insert a delay between the time the card goes off hook and the time it starts dialing? Alternatively, can I make it wait until there is a dial tone? Incoming calls are just fine, so I am almost certain this is what's happening. Thanks! Jim Asterisk/Zaptel has NEVER detected dial tone. Inserting multiple w's as others have mentioned only seems to work with Tone dialing as well, which is OK for 95% of the people. Of course you still have to guess at the number of w's, and as Cox gets slower, you will have to go back and insert yet another. No one seems to be interested or skilled enough to fix this. It's much more fun to add new wiz bang features than to fix some fundamental design flaws. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow to get dialtone when going off hook - bigproblem for me :(
- Original Message - From: John Novack [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 13, 2006 6:46 AM Subject: Re: [asterisk-users] Slow to get dialtone when going off hook - bigproblem for me :( Jim Archer wrote: Hi All... My Asterisk system uses VoIP and also 2 POTS lines from Cox Communications. Recently, the dial tone presentation from Cox seems to have slowed, so it can take as long as 3 seconds to get a dial tone. The problem I am having is that Asterisk does not seem to wait for the dial tone when dialing out. I'm using zaptel T400 cards. Is there any way to configure it such that I can insert a delay between the time the card goes off hook and the time it starts dialing? Alternatively, can I make it wait until there is a dial tone? Incoming calls are just fine, so I am almost certain this is what's happening. Thanks! Jim Asterisk/Zaptel has NEVER detected dial tone. Inserting multiple w's as others have mentioned only seems to work with Tone dialing as well, which is OK for 95% of the people. Of course you still have to guess at the number of w's, and as Cox gets slower, you will have to go back and insert yet another. No one seems to be interested or skilled enough to fix this. It's much more fun to add new wiz bang features than to fix some fundamental design flaws. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users How hard would it be to have asterisk detect a dial tone ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow to get dialtone when going off hook - bigproblem for me :(
Dovid B wrote: snip How hard would it be to have asterisk detect a dial tone ? I really can't say. I am not a C programmer, so I wouldn't even know where to start. Given that cheap dial up modems have, for the past ??20?? years, have been able to do just that, I would think it should have been an early consideration For those 1% of users, the last time I tried, the insertion of a w had no effect for pulse dialing either. JN ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk VM with Cisco routing
Has anyone out there implemented a system that does call routing via Cisco gear but VM for everyone on the system via Asterisk? What have been your successes and failures or issues? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headaches with Video over SIP
any logs/errors when you do a verbose 6 and a sip debug ?On 11/13/06, Peter Howard [EMAIL PROTECTED] wrote:On Mon, 2006-11-13 at 00:57 +0100, Patrick wrote: On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote: Greetings all, I'm playing with asterisk and two Polycom VSX300 videoconferencing units.And I'm having zero luck getting video working over SIP. The two units register fine with asterisk, and with allow=all in sip.conf, the two units establish voice.But no video.And no obvious messages as to whats going wrong.The config for each is (they're numbered 201 and 202): [202] secret= type=friend context=from-sip-202 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=all If you're wondering why I do the disallow=all immediately followed by allow=all, it's because the allow line has spent a lot of time with restricted codecs to see if that makes a difference. I can provide the full sip.conf, extensions.conf, and debug output if anyone wants to see them. Any suggestions as to where things are falling down? Do you have videosupport=yes in your sip.conf?Yes I do.I've also confirmed that I have a version of asterisk whichincludes the patch for H263P (which is what the Polycoms want to talk). --Peter HowardURSYS13 Burwood Rd,Burwood, NSW 2134Ph: 02 8745 2816Fax: 02 8745 2828___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Slow to get dialtone when going off hook - bigproblem for me :(
--On Sunday, November 12, 2006 11:53 PM -0500 John Novack [EMAIL PROTECTED] wrote: Dovid B wrote: snip How hard would it be to have asterisk detect a dial tone ? I really can't say. I am not a C programmer, so I wouldn't even know where to start. Given that cheap dial up modems have, for the past ??20?? years, have been able to do just that, I would think it should have been an early consideration For those 1% of users, the last time I tried, the insertion of a w had no effect for pulse dialing either. Well thanks to everyone who responded, and thanks to multiple w's I am back in operation. I went off hook a bunch of times and the worst case seemed to be 3 seconds to get a dial tone (which is pretty bad). It's hard to google one letter, but I eventually found that each w is .5 seconds, so 7 w's were inserted to be safe. I also called Cox and griped but I doubt that will do me any good. I am a C programmer, but I don't know anything about the inards of Asterisk. However, I would expect that dial tone detection would be a function of the hardware, not the Asterisk software. The cheap modems do this on board and export a simple command set. But I also don't know anything about Digium's hardware either. Thanks again! I really appreciate the help! Jim ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with CDR interpretation
Hello, About my problem with CDR where 2 calls overlaps, and there is no evidence that 3 other calls failed: after some searching on Asterisk bugs database I have found: http://bugs.digium.com/view.php?id=6762 ... When the Attended Transfer is used the information for call duration and who is talking is missed. ... I think this is the same problem as mine and there is no solution yet :( I will have to search for some workaround. Any ideas? Regards, Michał Niklas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headaches with Video over SIP
On Mon, 2006-11-13 at 13:42 +0800, Rosli Sukri wrote: any logs/errors when you do a verbose 6 and a sip debug ? I've been running with verbose 9, debug 9, and sip debug. The resultant output seems fine. The only warning is: WARNING[6964]: chan_sip.c:3592 process_sdp: Unknown SDP media type in offer: data 49218 RTP/AVP 100 The rest of the output seems to be normal. I can regenerate it, but right now I've put 1.4-beta3 on to see if that improves things (so far it hasn't, but I've tried one run) On 11/13/06, Peter Howard [EMAIL PROTECTED] wrote: On Mon, 2006-11-13 at 00:57 +0100, Patrick wrote: On Mon, 2006-11-13 at 10:45 +1100, Peter Howard wrote: Greetings all, I'm playing with asterisk and two Polycom VSX300 videoconferencing units. And I'm having zero luck getting video working over SIP. The two units register fine with asterisk, and with allow=all in sip.conf, the two units establish voice. But no video. And no obvious messages as to whats going wrong. The config for each is (they're numbered 201 and 202): [202] secret= type=friend context=from-sip-202 host=dynamic nat=no canreinvite=yes dtmfmode=rfc2833 disallow=all allow=all If you're wondering why I do the disallow=all immediately followed by allow=all, it's because the allow line has spent a lot of time with restricted codecs to see if that makes a difference. I can provide the full sip.conf, extensions.conf, and debug output if anyone wants to see them. Any suggestions as to where things are falling down? Do you have videosupport=yes in your sip.conf? Yes I do. I've also confirmed that I have a version of asterisk which includes the patch for H263P (which is what the Polycoms want to talk). -- Peter Howard URSYS 13 Burwood Rd, Burwood, NSW 2134 Ph: 02 8745 2816Fax: 02 8745 2828 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Howard URSYS 13 Burwood Rd, Burwood, NSW 2134 Ph: 02 8745 2816Fax: 02 8745 2828 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel compile problems
On Sun, Nov 12, 2006 at 07:26:02PM -0600, Traue, Paul wrote: I'm having difficulties getting zaptel to compile. I've compiled it in the past and never had any difficulties to speak of, but on this particular machine I have problems. The OS configuration is the same as I've used in the past and the hardware is identical. Obviously there's some subtle difference (probably with the packages installed, but I haven't been able to locate anything). Everything works fine until I get to the xpp stuff: CC [M] /usr/src/zaptel-1.2.6/ztdummy.o CC [M] /usr/src/zaptel-1.2.6/xpp/card_fxs.o /usr/src/zaptel-1.2.6/xpp/card_fxs.c: In function `FXS_card_new': /usr/src/zaptel-1.2.6/xpp/card_fxs.c:98: error: invalid use of undefined type `struct module' Fixed in later version. Get the latest 1.2 version (should be safe on Zaptel, as it does not change much). Or use latest debs (currently still 1.2.10, I know) Else, disable the building of that directory. There's a 'obj-m += xpp' in the Makefile. If you do have xpp (Astribank) hardware you definetly need a newer version as anything before 1.2.7 is a very old version of our driver. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Moh stops immediately
I'm trying to set up the Music on Hold feature. However, when I place a call the moh starts and stops immediately and as a result I dont hear the audio. -- Started music on hold, class 'default', on channel 'SIP/XXX' -- Stopped music on hold on SIP/XXX NOTICE[380]: res_musiconhold.c:515 monmp3thread: Request to schedule in the past?!?! My extensions.conf reads: exten = 2000,1,Answer exten = 2000,2,MusicOnHold(default) I've also tried: exten = 2000,1,Answer exten = 2000,2,MusicOnHold(default) exten = 2000,3,WaitMusicOnHold(20) exten = 2000,4,Hangup Any ideas? Thanks, zen perry Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail. http://new.mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can i have two asterisk vcersions running on same PC??
Can two versions of asterisk run on same PC?? Keerthy, Tr. Software Engineer, PrimeSoft IP Solutions Pvt. Ltd., Ph : 9246281937 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can i have two asterisk versions running on same PC??
Can two versions of asterisk run on same PC?? Thanks in advance, Keerthy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
