Re: [Asterisk-Users] press # to execute
Read takes the Digits that they enter, and puts it into a variable. Then I can take that variable and put it into the dial() command. exten = 877XXX,1,ANSWER exten = 877XXX,2,DigitTimeout,5 exten = 877XXX,3,ResponseTimeout,15 exten = 877XXX,4,Read(Secret,IVR/en_enter_destination,0) exten = 877XXX,5,dial(SIP/[EMAIL PROTECTED]) This is what I have. So When someone calls into the 877 DID, they hear enter destination, then they can enter in the phone number they wish to call, press # to execute it, so they don't have to wait for the timeout, which is done by the read() cmd. And then it sends the call to my LD provider. On Sun, 7 Nov 2004 09:49:53 -0800 (PST), oi geli [EMAIL PROTECTED] wrote: Mike, Please elaborate it little bit. I am having the same problem. Are you using read() for the EXTEN variable? if so, how? Thanks I found it, read() does exactly what I need On Sun, 7 Nov 2004 06:09:51 -0800, Mike Roberts manipura at gmail.com wrote: I'm trying to do this from PSTN - DID - * And yes, please spare me the lecture of security, I already know. On Sun, 7 Nov 2004 09:01:57 -0500, Steve Totaro asterisk at totarotechnologies.com wrote: That would be implimented on the phone. Grandstream is like that but on the snom you press OK. - Original Message - From: Mike Roberts manipura at gmail.com To: asterisk-users at lists.digium.com Sent: Sunday, November 07, 2004 7:08 AM Subject: [Asterisk-Users] press # to execute I have this. exten = 8,1,ANSWER exten = 8,2,DigitTimeout,5 exten = 8,3,ResponseTimeout,10 exten = 8,4,playback(IVR/en_enter_destination) exten = _1XXX.,1,dial(SIP/[EMAIL PROTECTED]) Basicaly its like pressing 8 for long distance, but more controled. But it has to wait until the timeout before it starts to dial. Is there a way to make them press # when they are done dialing the num in order to execute the _1XXX. I want to turn the timeout up but don't want to have them waiting forever. I also need to have a exten = _011. in there as well. So it won't have the same amount of digits everytime. ___ Asterisk-Users mailing list Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 500 software?
Polycom ships out two different phones , ones with H323,and one with SIP already loaded. Thank you, Steve Maroney Correction, the polycom IP 500 ships without h.323 or SIP software (it only has a bootrom on it), and software is only distributed by polycom authorized VoIP partners. I have personally taken issue with this as they advertise the product as H.323 and SIP compliant, yet without additional software it does not even know what SIP or H.323 are. That's not right. New phones come loaded with the current relevant firmware. Upgraded f/w is only available to/from certified resellers. Or look on the wiki for where it is freely available. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Dev] Illegal Instruction (Solved)
Hi all, I solved the 'Illegal Instruction' problem. This is what i did, hope this might help someone later... From my /proc/cpuinfo file: model name : VIA Samuel 2 I found this entry in the Asterisk Makefile and uncommented it: # Pentium VIA processors optimize # PROC=i586 Recompiled.. and now everything is OK. -Girish (Happy) --- Girish Gopinath [EMAIL PROTECTED] wrote: Hi Matt, --- Matt Gibson [EMAIL PROTECTED] wrote: did you delete your old asterisk's modules directory and try again? Thanks for the response. Yes, I removed everything before installing. Including /usr/lib/asterisk/modules and all directories under /var/lib/asterisk. But no luck. -Girish Sorry for the cross-post. I posted this to the -users list about 12 hours back and havent got any reply. Probably nobody there had experienced this problem. Can someone take a look into this and tell me why Asterisk seg-faults? --- Girish Gopinath [EMAIL PROTECTED] wrote: Folks, I have an RH machine which was running Asterisk 1.0-RC1. This evening i switched to Asterisk 1.0.1. Installation was successful, however Asterisk terminates abnormally during startup flashing an 'Illegal Instruction' message on the console. I noticed that this happens while loading the iax2 module. I am attaching the trace of core with this. Can anyone tell me what is going wrong and how to fix it? TIA, Girish From the console: [EMAIL PROTECTED] asterisk-1.0.1]# asterisk -cvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk 1.0.1, Copyright (C) 1999-2004 Digium. . . [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == IAX Ready and Listening on 0.0.0.0 port 4569 Illegal instruction (core dumped) __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astricon Brazil. Why not ?!
Hello list , I'm looking for partners in Brazil to discuss a possible way to have in Brazil an Official Conference regards Asterisk. It'll includes a hardware/workshop and tech-seminars. Would be nice if we could include in this conference , Anatel's presence and a seminar about the lawful aspects of VoIP in Brazil. I'm 100% sure that in Brazil , we have enough resources to become a large and active Asterisk community. :) Best Regards, -Jefferson Carvalho Jeff Networks Consulting Ltda. Teresina-PI-Brazil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip.conf extensions.conf
Title: Re: [Asterisk-Users] sip.conf extensions.conf
Hi, my sip.conf and my extensions.conf :)
I hope it's useful
**SIP.CONF**
[general]
port = 5060 ; port to bind for sip connections
bindaddr = 0.0.0.0 ; ip to bind for sip connections
context = default ; default context for incoming sip calls
externip = 222.99.99.22 ; Your external ip
localnet = 192.168.1.0/255.255.255.0 ;localnet and mask
disallow = all ; disallow all codecs, we want to enable,
allow=g726
allow=ulaw
allow=alaw
allow= gsm ; what we deem is necessary
allow= ilbc
allow= speex
register =
sipphonenumber:[EMAIL PROTECTED]/marlow-sip ;information
about sipphone
[proxy01.sipphone.com]
type=friend
username=sipphonenumber
secret=sipphonepwd
host=proxy01.sipphone.com
context=sipphone
nat=1
[marlow]
callerid=(marlow 3986)
username=marlow
type=friend
secret=marlowpwd
host=dynamic
context=internal
canreinvite=no
nat=1
[brandon]
callerid=(brandon 3986)
username=brandon
type=friend
secret=brandonpwd
host=dynamic
context=internal
canreinvite=no
[david]
callerid=(david 3988)
username=david
type=friend
secret=davidpwd
host=dynamic
context=internal
canreinvite=no
---
**EXTENSIONS.CONF**
[general]
static=yes
writeprotect=no
[globals]
MARLOW_CID=brandon
MARLOW_SIPPHONE=sipphonenumber
PHONE1=SIP/marlow ;unuseful for now it's only a try
PHONE2=SIP/brandon ;unuseful for now it's only a try
PHONE3=SIP/david ;unuseful for now it's only a try
[internal]
include = from-sip
include = sipphone
include = tollfree
include = 3986
include = 3987
include = 3988
include = voicesystem
[voicesystem]
exten = ,1,VoiceMailMain(${CALLERIDNUM}) ; extension is the VM
system,go directly to callers VM
exten = ,2,Hangup
[3986]
exten = 3986,1,Dial(SIP/marlow,20) ; call SIP extension marlow
for 60 seconds,if extension 3986 is called
exten = 3986,2,Voicemail(u3986) ; if we can't connect to
marlow or after seconds go to the unavail VM
exten = 3986,102,Voicemail(b3986) ; if busy, go to the busy VM
[3987]
exten = 3987,1,Dial(SIP/brandon,60) ; call SIP extension
brandon for 60 seconds,if extension 3986 is called
exten = 3987,2,Voicemail(u3986) ; if we can't connect to
brandon or after seconds go to the unavail VM
exten = 3987,102,Voicemail(b3986) ; if busy, go to the busy VM
[3988]
exten = 3988,1,Dial(SIP/brandon,60) ; call SIP extension david
for 60 seconds,if extension 3986 is called
exten = 3988,2,Voicemail(u3986) ; if we can't connect to
david or after seconds go to the unavail VM
exten = 3988,102,Voicemail(b3986) ; if busy, go to the busy VM
[from-sip]
;
; default extension for calls from SIP
;
; calls from sipphone
;for receive call from sipphone and send it to local phone 3986 but don't
work:( and I don't know why
exten = marlow-sip,1,SetCIDName(SIP - ${CALLERIDNAME}) ; indicate that the
call came through sipphone
exten = marlow-sip,2,Dial(Local/[EMAIL PROTECTED]/n)
[outbound-internal]
;
; include local extensions
;
include = internal
;
; include SIP accounts
;
include = sipphone
;
; include tollfree calls
;
;include = tollfree
[default]
; include from-sip for default. We don't use it, but it might be a good idea
include = from-sip
include = sipphone
include = internal
[sipphone]
; Official Sipphone example don't work very well
; exten = _1747.,1,Dial(SIP/[EMAIL PROTECTED]) ; set my
callerid and name
; exten = _1747.,2,Playback(notavail) ; this did
not work out
; exten = _1747.,3,Busy
;Approach to gateway guide
exten = _1747.,1,SetCallerID(${MARLOW_CID} ${MARLOW_SIPPHONE}) ; set my
callerid and name
exten =
_1747.,2,Dial,SIP/${EXTEN:[EMAIL PROTECTED] ; dial
the number i wish to dial
exten = _1747.,3,Playback(invalid) ; this did
not work out
exten = _1747.,4,Hangup
exten = _1747.,103,Busy
[tollfree]
;
; terminate toll-free no.'s via fwdnet
;
;Use for call italian toll free
; +39 800
; exten = _39800.,1,SetCallerID(${MARLOW_SIPPHONE})
; exten = _39800.,1,Dial,SIP/[EMAIL PROTECTED]
; exten = _39800N.,1,Dial,Zap/1/${EXTEN:2}
; Use for call external PSTN number
exten = _0X.,1,Dial,Zap/1/${EXTEN:1}
exten = _0X.,2,Playback(invalid)
exten = _0X.,3,Hangup
exten = _0X.,103,Busy
;Use for call american Toll free
; +1-800
exten = _1800.,1,SetCallerID(${MARLOW_SIPPHONE})
exten = _1800.,2,Dial,SIP/[EMAIL PROTECTED]
exten = _1800.,3,Playback(invalid)
exten = _1800.,4,Hangup
exten = _1800.,103,Busy
---
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[Asterisk-Users] Have anyone try to use asterisk as a business mode
hi everybody, I have study * recently,as i see,* is suit to build a small ip pbx. Have anyone try to porting it to a embeded linux box,and sale as a small ip-pbx? Do you think this business module is possible ?Do You Yahoo!? 150MP3 1G1000___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 1751-V SIP Gateway for Asterisk
I have a 1751 with a BRI Wic, I would like it to pass
incoming calls to Asterisk.
After spending a lot of time on this, I cannot get it to
work. I can see the incoming call and the callerID, yet the router
doesnt seem to pass the call to asterisk.
In SIP.conf
[213.137.185.150]
context=incoming
type=friend
host=213.137.185.150
dtmfmode=rfc2833
disallow=all
allow=ulaw
allow=alaw
In extensions.conf incoming context:
123456789 is an example of our phone number.
exten = 123456789,1,Wait(1)
exten = 123456789,2,Dial(SIP/5011,15)
exten = 123456789,3,VoiceMail(u${5011})
exten = 123456789,4,Congestion
exten = 123456789,102,Hangup
Can anyone provide me a working config with BRI and a 1751.
We are in UK.
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Re: [Asterisk-Users] SMS through Cisco PSTN GW
Nahuel Alejandro Ramos wrote: Hi everyone, I have my asterisk working with a Cisco 2610 PSTN Gateway connected over SIP protocol. Could anybody tell me if I can send and receive SMS through this Gateway with the SMS command in asterisk? Depends on the codec really. Landline SMS is sent via FSK modulation. I'm guessing that if you're using ulaw/alaw codecs for the call you shoudln't have a problem. You might also have to shut off echo cancelation. Cheers, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID Name from SIP to IAX2
Hi,
- Original Message -
From: Dan [EMAIL PROTECTED]
I have upgraded the Asterisk Server after several months and now there is
an issue with the CallerID Name information.
When I call from DIAX (IAX2) to ATA186(SIP) I get the correct CallerID
name/number.
When I call from ATA186(SIP) to DIAX(IAX2) I get the correct number, but
the CallerID Name is unknown
Checking the ${CALLERIDNAME} variable, it is ok in the first case and
empty in the second one.
In sip.conf I have the line:
callerid=Namenumber
as before.
There is something changed in the Asterisk Server in between, related to
the CallerID Name information?
I'm back with some more info:
It works with
Asterisk CVS-12/12/03-11:11:35
but doesn't ($CALLERIDNAME} is empty) with
Asterisk CVS-HEAD-10/24/04-21:13:25
Same config files are used in both situation.
I must change something?
Thanks,
Dan
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[Asterisk-Users] re: CallerID for the UK
Hi Guys, Hi all, I am too new with Linux, to really experiment with the callerid. I know the problem is due to BT using a different technical platform. So as too using Perl or any other scripts and a modem to create the work around required to get the callerid to work UK I need some help in this field. Anyone, got a step-by-step guide, how to add this to a working setup? This would solve my last technical issue. Many thanks in advance also any one got any RPMs for a GUI which can be used to setup SIP and IAX account and configure the dial plan through a nice web interface. Hoping to have helped, Charles Osstyn 11, Cowper Crescent Foxhill, Sheffield S6 1AU United Kingdom Standard contact channels Tel +44 (0)114 231 38 98 (Now connected to our VOIP server Gonzo, hit option five for my office line.) Mob +44 (0)790 393 91 46 Fax +44 (0)870 051 79 92 Preferred VOIP contact channels SIP sip:[EMAIL PROTECTED] (Get a free pre-configured (with account) VOIP soft phone from FWD (Free World Dialup) here for your PC, laptop or Pocket PC.) E-mail [EMAIL PROTECTED] Web www.osstyn.com Webcam www.osstyn.com:81/guest.htm On request via Skype. Skype charelke (Get the free Skype VOIP client here.) MSN Messenger [EMAIL PROTECTED] (Get MSN Messenger here.) E-MAIL DISCLAIMER The information in this e-mail is confidential, and is intended solely for the addressee's. Access to this email by anyone else is unauthorised and therefore prohibited. If you are not the intended recipient, or if the email is marked as 'confidential', any disclosure, copying, distribution or any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. BEGIN:VCARD VERSION:2.1 N:Osstyn;Charles;;Mr. FN:Charles Osstyn ([EMAIL PROTECTED]) TITLE:Business Analyst E-Consultant TEL;HOME;VOICE:+44 (114) 2313898 TEL;CELL;VOICE:+44 (790) 3939146 TEL;HOME;FAX:+44 (870) 0517992 ADR;WORK:;;11, Cowper Crescent;Foxhill, Sheffield;South-Yorkshire;S6 1AU;United Kingdom LABEL;WORK;ENCODING=QUOTED-PRINTABLE:11, Cowper Crescent=0D=0AFoxhill, Sheffield, South-Yorkshire S6 1AU=0D=0AUni= ted Kingdom ADR;HOME:;;11, Cowper Crescent;Foxhill, Sheffield;South-Yorkshire;S6 1AU;United Kingdom LABEL;HOME;ENCODING=QUOTED-PRINTABLE:11, Cowper Crescent=0D=0AFoxhill, Sheffield, South-Yorkshire S6 1AU=0D=0AUni= ted Kingdom URL;WORK:http://www.osstyn.no-ip.com EMAIL;PREF;INTERNET:[EMAIL PROTECTED] REV:20031214T060447Z END:VCARD ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: SIP REGISTER -- Asterisk non-compliant or is it the provider?
On Mon, 08 Nov 2004 07:03:49 +0100, Tom Ivar Helbekkmo [EMAIL PROTECTED] wrote: Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] writes: You found the right one. Here's what was posted to the CVS list: [SNIP] Setup fromuser properly (bug #2802) No, I did not forget that one. I was the one who reported it in the first place. Instead, it would seem that you didn't read my post carefully ;-) Actually, Benjamin, it would seem *you* did not read *my* post Indeed, and I offer my apologies. I swear I remember to have seen forget there instead of found. Then I thought it was a response to the latest post in which I posted my alternative fix. It seems my mind was already in a meeting I was about to leave for. I am sorry about this. thanks for the reply rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] missing wakeup gsm files
Where can I download the missing wakeup gsm files? These are in the Asterisk sounds addon CVS. Very well documented on the Wiki. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CVS RPMs for Mandrake 10 (Zaptel and, Asterisk)
On Sunday 07 November 2004 07:39 am, Clive Carter wrote: Dear Scott I am new user of Mandrake 10 And very excited at the idea to work with Asterisk but, as you can imagine. I am currently blocked because of the kernel 2.6.. the Wildcard X100P drivers . I would be more than happy to get test your source RPMs for zaptel and asterisk And so would I !! Don't know what block you are talking about. MDK 10 the 2.6 kernel works great with X100P. (Unless something recently has changed.) -- Steve Szmidt They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No busy-tone
Just like this? It doesn't seem to work though. [wx3trunk-outgoing] include = internal-sip-callers exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,T) exten = s,101,Busy exten = h,1,Hangup Nicklas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: den 7 november 2004 23:29 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No busy-tone Oh that! Just put a Busy() at priority+101 Look at the [macro-std-exten] in the asterisk/configs/extensions.conf.sample for another way to do it. The remote device is telling Asterisk the destination is busy. Now you have to tell your dialplan what to do. Do you send a busy tone to the caller? Do you dial a different destination? Nicklas Bondesson wrote: Ok, here't the output from console: Executing Dial(SIP/200-f359, SIP/[EMAIL PROTECTED]||T) in new stack -- Called [EMAIL PROTECTED] -- Got SIP response 486 Busy here back from xxx.xxx.xxx.xxx -- SIP/xxx.xx-7d58 is busy == Everyone is busy/congested at this time Thanks Nicklas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: den 7 november 2004 20:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No busy-tone Nicklas Bondesson wrote: I don't hear anything. There's no sound at all. Nicklas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: den 7 november 2004 17:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No busy-tone Nicklas Bondesson wrote: I don't hear a busy-tone when calling an external extension that's busy. I just get the Busy Here 486 message in the debugging log. Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward incoming SIP calls to H323 ipphone?
Since you are able to receive H.323 calls with chan_oh323, I assume that the module is loaded. You could check the incoming/outgoing/simultaneous limits or submit the oh323.conf. Additionally, what are the full messages that you get on the console? Michael. Alex van Es wrote: Hi all, For my setup I need to forward incoming SIP and ZAP calls to my IP phone using H323. I have managed to set up the OH323 and when I enter my asterisk's ip number into sjphone, it will answer and give me the welcome message. So receiving calls with H323 is not a problem.. but I want to be able to dial out. I have set up a extention that looks like; exten = 1234,1,Dial(OH323/192.168.1.20) I keep on getting the message unable to create channel of type ' OH323'. I have tried also the names h323, h.323, oh323, OH323/h323.. but none of them seem to exist. When I receive the incoming call it says channel OH323, so I assume that is the correct name. However.. I still can't forward calls out. I could do without OH323, but when I forward incoming SIP calls to my IP phone using SIP I just get silence after I answer the phone (both parties can't hear each other) so I wanted to try it this way. Anyone has any ideas? Alex -- Alex van Es - [EMAIL PROTECTED] http://photography.icepick.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk and nat
Hi all, Hope somebody can help me to figure out the following scenario or send me on their config files if they have a similiar network configuration. I first set up asterisk and two clients on the same network and it worked fine. I now have asterisk set up which is acting as a sip registrar. It is behind nat. I also have two clients which are behind nat on two separate networks. I can no longer register the clients. I have set 'nat=yes' in the client config but is there something else I must do for the asterisk sevrer itself?... I find this situation confusing so if someone could clarify it for me, I would be very grateful. Thanks in advance, Aisling. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI Errors
I'm having troubles with my agi scripts. -- Executing Answer(SIP/asterisk-7f82, ) in new stack -- Executing AGI(SIP/asterisk-7f82, php-agi.agi) in new stack Failed to execute '/var/lib/asterisk/agi-bin/php-agi.agi': No such file or directory Now that file is there! Thats a fact. The permissions are right (I hope) and I pulled the script off a working server. I had a cvs, I updated to 1.0 (where the script came from) and still nothing. Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help on Supervised Call Transfer
Hy guys I'm new to Asterisk and I would appreciatesome help from you. I have a TDM400P board with 2 FXO and 2 FXS modules. I must implement an application that answer to a call made on an external line and enter in a AGI script (php) where there is an IVR menu. If the caller want to contactanother person on his mobile phone(let's say), then I must do a supervised call transfer(put the caller on hold, pick-up the otherexternal line, call that person and tell him that someone is looking for him; if he accept the call then I must link the two persons). All this I should make them from my php-agi script. If someone can give me some push on this issue I will be grateful. Regards Cristian Afilipoaie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Errors
On Mon, 2004-11-08 at 03:00 -0800, Mike Roberts wrote: I'm having troubles with my agi scripts. -- Executing Answer(SIP/asterisk-7f82, ) in new stack -- Executing AGI(SIP/asterisk-7f82, php-agi.agi) in new stack Failed to execute '/var/lib/asterisk/agi-bin/php-agi.agi': No such file or directory Now that file is there! Thats a fact. The permissions are right (I hope) and I pulled the script off a working server. I had a cvs, I updated to 1.0 (where the script came from) and still nothing. Any ideas? Not unless you provide some more details. Why don't you paste in a ls -l of the above quoted file with full path? Also verify the she-bang line. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_sccp and Cisco 7940
Answering myself (but might be useful to anyone else):
The Cisco 7940 / chan_sccp DOES answer calls just fine providing that I
only setup one line (the 7940 has two line buttons). So this is
getting me by with this phone perfectly although I'm sure I'm missing
something here as the multiple lines with chan_sccp should work and I
haven't seen anyone else ask this not-picking-up question - this should
be especially important for 7960s which have 6 line buttons.
The only thing that I find is a bit of a pity is that there doesn't seem
to be a way to make the Voicemail button work (i.e. automatically dial
the mailbox extension) - when the voicemail button is pressed this comes
up on the asterisk console: Got {StimulusMessage}
stimulus=VoiceMail(15) stimulusInstance=1 and, on the next line,
sccp_actions.c:343 sccp_handle_stimulus: VM Button is not yet handled.
working on implementation.
I think this wouldn't be such a problem with a 7960 because a speeddial
button could be used to easily access voicemail.
Derek
Derek Conniffe wrote:
Hi everyone,
I have a Cisco 7940 and I'm using chan_sccp with it (chan_skinny does
work fine but it seems to be very featureless compaired to chan_sccp -
caller Id being probably the biggest reason to use the latter).
I can make call on the 7940 but I cant answer them. The phone rings
but when I pick up the call the phone just keeps ringing. I can press
the Answer soft button but nothing happens and then when I hang the
phone up asterisk crashes with a Segmentation fault and the console
says the following:
Oct 29 11:48:15 ERROR[1108667312]: sccp_actions.c:449
sccp_handle_onhook: Erp, tried to hangup when we didn't have an active
channel?!
== Spawn extension (default, 6101, 2) exited non-zero on
'SIP/derekdesk-c963'
== Sending Packet Type KeepAliveAckMessage (4 bytes)
== Sending Packet Type SetLampMessage (16 bytes)
Segmentation fault (core dumped)
Does anyone have any idea how to fix this?
Thanks,
Derek
--
Derek Conniffe
Rivertower Ltd
DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146
Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823
Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180
Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085
Email: [EMAIL PROTECTED]
Web: www.rivertowerhosting.com
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Re: [Asterisk-Users] Cisco 7970 Firmware for the 7960G
I'm using a 7940 with the Call Manager firmware and chan_sccp to make the phone work with Asterisk. It mostly works ok [with the sccp channel] but I think that you'd be a goot bit better off with the SIP firmware but I haven't tried this myself because, like you've identified, you need to have some kind of contract to get access to the different firmware versions. Derek [EMAIL PROTECTED] wrote: Hello, i´m thinking about buying one if the Cisco´s CP-7970G Phone. Does someone can confirm that it will work with asterisk? I also have some trouble getting the newest firmware for my CP-7960G as Cisco doesn´t support people from outsite U.S. without a Support Contract(even with warranty) and it is very hard to get one here in Germany. Can someone please email me the latest upgrade for my two days old 7960G? :-) Regards Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Brazil. Why not ?!
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jefferson, Jefferson Carvalho wrote: | Hello list , | | I'm looking for partners in Brazil to discuss a possible way | to have in Brazil an Official Conference regards Asterisk. Yes, it will be wonderfull ! | It'll includes a hardware/workshop and tech-seminars. | Would be nice if we could include in this conference , Anatel's | presence and a seminar about the lawful aspects of VoIP in Brazil. | I'm 100% sure that in Brazil , we have enough resources to | become a large and active Asterisk community. :) I'm glad to say that you are right and more, there are a lot of people in Brazil working with Asterisk. Contact me in private if you want. Best regards. | | Best Regards, | | -Jefferson Carvalho | Jeff Networks Consulting Ltda. | Teresina-PI-Brazil | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http://www.devel-it.com.br TDKOM Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBj19BiLK8unYgEMQRAqG4AJ0azCrspMj2Ca0m/bc6FERBf2lP6QCfYKEZ oguuMN/B5xP8WofrQppKI6Y= =9dRq -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens GSM terminal with Wildcard FXO
I picked up a CellSocket nokia GSM phone to POTS adapter for about US$25 (although I'm in Ireland so it cost me more on courier charges). Its connected to an X100P and working very well for me. Derek Ronan Mullally wrote: On Sun, 7 Nov 2004, Martin List-Petersen wrote: or you can look at chan_bluetooth (http://www.crazygreek.co.uk/content/chan_bluetooth). It's a work in progress, but seems to do the job, if you can get audio working. A cellphone and a bluetooth module are usually quite a lot cheaper than a GSM-to-PSTN adapter (usually 500 EUR and up) and a FXO or FXS (100 EUR and up) device. You can get FTCs for well under 500 Euros - I can get my hands on Ericsson F151s for about 230 Euros + VAT, assuming reasonable (a dozen or so) quantities. -Ronan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Derek Conniffe Rivertower Ltd DDI: (Local Ireland) 01 201 0146 (International) +353 1 201 0146 Mobile: (Local Ireland) 086 856 3823 (International) +353 86 856 3823 Main Line: (Local Ireland) 1890 45 70 74 (International) +353 1 201 0180 Fax: (Local Ireland) 01 201 0085 (International) +353 1 201 0085 Email: [EMAIL PROTECTED] Web: www.rivertowerhosting.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Enhanced Audio Support for EAGIs
Hi, I'd just like to say that I'm interested in this thing. Do you intend to use Sphinx 4 ? Can sphinx use HTK HMM models files ? Please keep us posted on progress Regards, Robert. - Original Message - From: Jeff Maki [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, November 07, 2004 1:46 AM Subject: [Asterisk-Users] Enhanced Audio Support for EAGIs Hey everybody, I'm a graduate student at Carnegie Mellon, and I'm working on a project that wishes to leverage the Sphinx speech-recognition system (also developed at CMU) with asterisk. I see that the EAGI spec provides for an audio stream on fd 3 for this exact purpose, but I can't seem to get it to appear--every time I run my EAGI script (written in C, read()ing fd 3), I get Resource unavailable, and confirming with a bash script that lists /dev/fd, only 0, 1 and 2 appear--no 3. What gives? Can somebody help me get it to work? Thanks in advance! BTW, I already posted to the dev list, but I got no reply--sorry if this is kinda off topic... -Jeff. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: VoiceMailMain(sexten@context) doesn't work in CVS 11/03
Nevermind. Another post answered my problem. The dial plan was chaning and ALERT_INFO needs to be changed to _ALERT_INFO, thanks for all of your help On Fri, 5 Nov 2004 09:31:27 -0500, Matthew Marlowe [EMAIL PROTECTED] wrote: Can anyone else verify this or is it just me? -- MBM -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: CVS-HEAD-11/03/04-14:09:34 ALERT_INFO Doesn't Get Passed
Nevermind. Another post answered my problem. The dial plan was chaning and ALERT_INFO needs to be changed to _ALERT_INFO, thanks for all of your help On Sun, 7 Nov 2004 21:14:13 -0500, Matthew Marlowe [EMAIL PROTECTED] wrote: Anyone having this problem? On Thu, 4 Nov 2004 09:14:33 -0500, Matthew Marlowe [EMAIL PROTECTED] wrote: I saw a previous post about this but I can't find it, CVS-HEAD-11/03/04-14:09:34 does not pass ALERT_INFO to the phones. It used to work but has now stopped. I'm not a coder so I can't look through the code but someone mentioned ALERT_INFO does not exist in app_dial if I remember correctly. Anyone know anything about this? -- MBM -- MBM -- MBM ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting jitterbuffer in with iax
Hello everybody; I would like to know the parameters on which depend jitterbuffer in iax.conf. Is there some kind of formula to set the correct values? Thanks in advance for any help Lamine ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] press # to execute
You could pass the pound sign as a PLAR (Private Line Automatic Ringdown) code, with say a PIN number after that. The PLAR code is also called Automatic Dial when Off-Hook. On Sunday 07 November 2004 12:08 pm, Mike Roberts wrote: I have this. exten = 8,1,ANSWER exten = 8,2,DigitTimeout,5 exten = 8,3,ResponseTimeout,10 exten = 8,4,playback(IVR/en_enter_destination) exten = _1XXX.,1,dial(SIP/[EMAIL PROTECTED]) Basicaly its like pressing 8 for long distance, but more controled. But it has to wait until the timeout before it starts to dial. Is there a way to make them press # when they are done dialing the num in order to execute the _1XXX. I want to turn the timeout up but don't want to have them waiting forever. I also need to have a exten = _011. in there as well. So it won't have the same amount of digits everytime. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 500 software?
Polycom ships out two different phones , ones with H323,and one with SIP already loaded. Thank you, Steve Maroney Correction, the polycom IP 500 ships without h.323 or SIP software (it only has a bootrom on it), and software is only distributed by polycom authorized VoIP partners. I have personally taken issue with this as they advertise the product as H.323 and SIP compliant, yet without additional software it does not even know what SIP or H.323 are. That's not right. New phones come loaded with the current relevant firmware. Upgraded f/w is only available to/from certified resellers. Or look on the wiki for where it is freely available. The two new 500's that were purchased from a Polycom reseller actually came with no firmware installed at all; only the bootloader (or whatever its called). Someone on this list pointed me to a souce for downloading the sip image, and now I've got the phone running, but it won't register with *. Not sure what the registration problem is as yet, but doing a sip debug indicates the registration failure. I double checked the Auth UserID and Password and they appear to be correct. Seems others on the list have had the same issue, but I've not found any responses resolving the problem as yet. Anyone have any suggestions? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Errors
The Script http://pastebin.ca/1968 The File [EMAIL PROTECTED] agi-bin]# ll -rwxr-xr-x1 root root 1020 Nov 8 01:17 php-agi.agi [EMAIL PROTECTED] agi-bin]# pwd /var/lib/asterisk/agi-bin The Error *CLI -- Executing Answer(SIP/asterisk-6520, ) in new stack -- Executing AGI(SIP/asterisk-6520, php-agi.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/php-agi.agi Failed to execute '/var/lib/asterisk/agi-bin/php-agi.agi': No such file or directory -- AGI Script php-agi.agi completed, returning 0 -- Executing DigitTimeout(SIP/asterisk-6520, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(SIP/asterisk-6520, 15) in new stack -- Set Response Timeout to 15 -- Executing Read(SIP/asterisk-6520, Secret|IVR/en_enter_destination|0) in new stack -- Playing 'IVR/en_enter_destination' (language 'en') Extensions.conf [tf-did] exten = 877XXX,1,ANSWER exten = 877XXX,2,agi(php-agi.agi) exten = 877XXX,3,DigitTimeout,5 exten = 877XXX,4,ResponseTimeout,15 exten = 877XXX,5,Read(Secret,IVR/en_enter_destination,0) exten = 877XXX,6,dial(SIP/[EMAIL PROTECTED]) Hope that helps On Mon, 08 Nov 2004 05:16:24 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2004-11-08 at 03:00 -0800, Mike Roberts wrote: I'm having troubles with my agi scripts. -- Executing Answer(SIP/asterisk-7f82, ) in new stack -- Executing AGI(SIP/asterisk-7f82, php-agi.agi) in new stack Failed to execute '/var/lib/asterisk/agi-bin/php-agi.agi': No such file or directory Now that file is there! Thats a fact. The permissions are right (I hope) and I pulled the script off a working server. I had a cvs, I updated to 1.0 (where the script came from) and still nothing. Any ideas? Not unless you provide some more details. Why don't you paste in a ls -l of the above quoted file with full path? Also verify the she-bang line. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Clipping at start of call
On Sun, 07 Nov 2004 18:05:19 -0700, Michael Loftis [EMAIL PROTECTED] wrote: I've also experience clipping though with cisco SIP phones as well as occasionally when dialing into our IVR from my Vonage (Cisco ATA) VoIP line at home. The clipping at the start of a call was resolved by Cisco (finally!) in their 7.x SIP firmware releases. I actually upgraded all our 7940s today for this very reason. -- Sam Bashton ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: CVS-HEAD-11/03/04-14:09:34 ALERT_INFO Doesn'tGet Passed
- Original Message - From: Matthew Marlowe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, November 08, 2004 7:32 AM Subject: [Asterisk-Users] Re: CVS-HEAD-11/03/04-14:09:34 ALERT_INFO Doesn'tGet Passed | Nevermind. Another post answered my problem. The dial plan was chaning | and ALERT_INFO needs to be changed to _ALERT_INFO, thanks for all of | your help | does this mean going forward, asterisk is no longer backward compatible with earlier versions of the dial plan greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quintum vs Asterisk
Hello Has anyone ever got a quintum A800 or A400 with SIP firmware on it succesfully talking to asterisk's SIP stack? I tried it.. But get many call leg does not exist errors Seems like quintum's sip implementation is not the most compatible one..?? Anyone experienced with this? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Siemens GSM terminal with Wildcard FXO
Well now that you mention it, I have Used a Telular GSM Fixed Wireless Terminal using asterisk to both place and receive bulk calls. I was using it to test the unit out, though I personnally wouldn't suggest the terminal ( seems like it is very buggy) it does seem to work. There are several options as it goes depending on the scale of your implementation. One was to look at it would be to use a FWT to connect to your FXO card or a T1 via a channelbank, a good solution but very clunky. one of the more clever ideas is the Chan_bluetooth which would wirelessly connect to your phone. I am interested in the possibility of connecting multiple phones using this method. If you are interested in high Density, There are GSM to T1/E1 channel banks out there but I am just begining to explore them. Please keep me informed of any success you may have in this as I am very interested. I work for a new GSM 850 provider here in the states. Chad C. Wicker Systems Engineer Petrocom [EMAIL PROTECTED] 11/7/2004 10:36:24 AM Hi Jafar, You want to look at Fixed Cellular Telephones (FCTs) like the Nokia Premicell or the Ericsson F251. As I understand it these present a PSTN interface which you can plug into an FXS interface. -Ronan On Sun, 7 Nov 2004, jafar mohammed wrote: Hi, I would like to implement GSM origination for a VOIP system i am developing. I am purchasing a Siemens M20 Terminal and would like to know if i can plug it into my Wildcard FXO device to get incoming GSM calls routed to the Asterisk server. If anyone has been able or successful in using this terminal please let me know. And if any of you have this terminal can you hook it up to a telephone headset and see if incoming calls will ring the headset. Thank you. __ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom 500 software?
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Peter Johnson Sent: Monday, November 08, 2004 1:02 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom 500 software? Polycom ships out two different phones , ones with H323,and one with SIP already loaded. Thank you, Steve Maroney Correction, the polycom IP 500 ships without h.323 or SIP software (it only has a bootrom on it), and software is only distributed by polycom authorized VoIP partners. I have personally taken issue with this as they advertise the product as H.323 and SIP compliant, yet without additional software it does not even know what SIP or H.323 are. That's not right. New phones come loaded with the current relevant firmware. Upgraded f/w is only available to/from certified resellers. Or look on the wiki for where it is freely available. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Are you sure your statement applies to the IP 500, the two I just RMA'd 30 days ago also did not come with any protocol application. At that time Polycom confirmed that they do not ship that model with a protocol application. With the IP 500 there is a difference between the firmware, boot loader, and application. Is it possible in you experience that the dealer or distributor loaded the firmware before shipping? The normal process for the IP 500 is to download the application you want to run (h.323 - aka HMX, SIP, or MCGP) to the phone from a user configured FTP server at boot time. If polycom has changed this that would be useful to know for sure. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Strategy in App_queue
On Mon, 08 Nov 2004 15:05:33 +0800, el Flynn [EMAIL PROTECTED] wrote: Nathan Bowyer wrote: Doesn't seem to work for me that way. Anyone else got any ideas? When I look at the code, it looks like copying what roundrobin does, then simply removing the pos whenever you complete a call (or one abandons) would reset the queue back to its original state. I can't seem to accomplish this, though. What about assigning penalties to the agents? The agent to call first would have the lowest penalty, increasing as you add agents to the list. Flynn While it does put them in the correct order this way, it seems to have a hard time progressing in penalties. Phone one will ring many many times, and if no one answers it will simply keep ringing. I suppose I could play with the metrics and penalties, making the second ring place the second phone as the lowest metric (the phone to be called). I'll have to check that out. Is anyone interested in something like this, or is this a change I should just keep to myself? :) Thanks, Nathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and nat
Ashling O'Driscoll [EMAIL PROTECTED] writes: I first set up asterisk and two clients on the same network and it worked fine. I now have asterisk set up which is acting as a sip registrar. It is behind nat. I also have two clients which are behind nat on two separate networks. I can no longer register the clients. I have set 'nat=yes' in the client config but is there something else I must do for the asterisk sevrer itself?... I am only able to give you a link which might help you. Benjamin on Asterisk always writes good articles about nat (and other things as well). Here is one of them: http://lists.digium.com/pipermail/asterisk-users/2004-October/068275.html You probably should read the whole thread. Click the link thread on that page and search in the new page for postings with the the subject Almost there--Remote connection. -- Fabian Müller ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 220 (or other phones) - line apperances?
Message: 11 Date: Mon, 8 Nov 2004 11:17:03 +1000 From: JB Hewit [EMAIL PROTECTED] Subject: [Asterisk-Users] Snom 220 (or other phones) - line apperances? To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII Hi, I've googled and searched the wiki but I'm not sure if it's entirely possible or not. I'm looking for 'line apperances' functionality with phones like the Snom 220. Essentially I wish to have buttons on a panel (like the Snom 220's extension board) that show when people are on the phone or off the phone for a receptionist. As far as I know, you can't do this with asterisk, at least not easily. From what I've read, most people call this shared lines or something similar. I've heard that MGCP does support something similar to this, but that Asterisk does not specifically support it. All of the line appearances on a multi-line sip phone are unique to just that phone - nobody on another phone can see them. This has obvious benefits over shared lines, but it sure leaves out operator monitoring of all the lines. Somebody, though, was kind enough to direct me to: http://www.asternic.org/ It is flash-based browser access to all your lines and extensions with cool drag and drop capabilities and some other nifty features. The other thing sorely missing from unique SIP lines, IMHO, is easy call parking. Instead of just pressing the hold button, you have to dial a specific extension to park a call. Of course, most sip phones have programmable buttons that allow you to do this in one key press. All in all, I'd still rather have the unique SIP lines rather than the shared lines. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Free World Dialup via IAX2 gives duplicate calls?
Hi: I have this problem trying to connect two asterisk servers via Free World Dialup's IAX2 (FWD) mechanism: Calls from one asterisk server seem to get duplicated when they get to the other asterisk server. This causes the extension to which the call is directed to appear busy causing the second call (which appears to be the real one) to be directed to voicemail. Has anyone else experienced this? Here are logs for an example of this: -- Accepting AUTHENTICATED call from 65.39.205.121, requested format = 4, actual format = 4 -- Executing Macro([EMAIL PROTECTED]/5, con-ext-cid|101|89Chris Hobbs) in new stack -- Executing SetVar([EMAIL PROTECTED]/5, name=Chris Hobbs) in new stack -- Executing SetCallerID([EMAIL PROTECTED]/5, 89Chris Hobbs) in new stack -- Executing SetCIDName([EMAIL PROTECTED]/5, Chris Hobbs) in new stack -- Executing Goto([EMAIL PROTECTED]/5, ext-int|101|1) in new stack -- Goto (ext-int,101,1) == Channel '[EMAIL PROTECTED]/5' jumping out of macro 'con-ext-cid' -- Executing Macro([EMAIL PROTECTED]/5, dial-ext-vm|SIP/101) in new stack -- Executing Dial([EMAIL PROTECTED]/5, SIP/101|15|t) in new stack -- Called 101 -- SIP/101-7d91 is ringing -- Accepting AUTHENTICATED call from 65.39.205.121, requested format = 4, actual format = 4 -- Executing Macro([EMAIL PROTECTED]/6, con-ext-cid|101|89Chris Hobbs) in new stack -- Executing SetVar([EMAIL PROTECTED]/6, name=Chris Hobbs) in new stack -- Executing SetCallerID([EMAIL PROTECTED]/6, 89Chris Hobbs) in new stack -- Executing SetCIDName([EMAIL PROTECTED]/6, Chris Hobbs) in new stack -- Executing Goto([EMAIL PROTECTED]/6, ext-int|101|1) in new stack -- Goto (ext-int,101,1) == Channel '[EMAIL PROTECTED]/6' jumping out of macro 'con-ext-cid' -- Executing Macro([EMAIL PROTECTED]/6, dial-ext-vm|SIP/101) in new stack -- Executing Dial([EMAIL PROTECTED]/6, SIP/101|15|t) in new stack -- Called 101 -- Got SIP response 486 Busy back from 192.124.97.45 -- SIP/101-153f is busy == Everyone is busy at this time -- Executing Ringing([EMAIL PROTECTED]/6, ) in new stack -- Executing Wait([EMAIL PROTECTED]/6, 1) in new stack -- Executing VoiceMail([EMAIL PROTECTED]/6, b101) in new stack -- Playing 'vm-theperson' (language 'en') == Spawn extension (macro-dial-ext-vm, s, 1) exited non-zero on '[EMAIL PROTECTED]/5' in macro 'dial-ext-vm' == Spawn extension (ext-int, 101, 1) exited non-zero on '[EMAIL PROTECTED]/5' -- Hungup '[EMAIL PROTECTED]/5' -- Playing 'digits/1' (language 'en') -- Playing 'digits/0' (language 'en') -- Playing 'digits/1' (language 'en') -- Playing 'vm-isonphone' (language 'en') == Spawn extension (macro-dial-ext-vm, s, 104) exited non-zero on '[EMAIL PROTECTED]/6' in macro 'dial-ext-vm' == Spawn extension (ext-int, 101, 1) exited non-zero on '[EMAIL PROTECTED]/6' -- Hungup '[EMAIL PROTECTED]/6' -- Robert Withrow, R.W. Withrow Associates, Swampscott MA, [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No busy-tone
Nicklas Bondesson wrote: Just like this? It doesn't seem to work though. [wx3trunk-outgoing] include = internal-sip-callers exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,T) exten = _X.,101,Busy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk and nat
thanks a million for the reply and link (even though as far as asterisk behind nat and sip goes it doesnt look promising)... Has anyone else any other ideas? Thanks again, Aisling. Original Message From: [EMAIL PROTECTED] (Fabian Müller) To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] asterisk and nat Date: Mon, 08 Nov 2004 16:40:32 +0100 Ashling O'Driscoll [EMAIL PROTECTED] writes: I first set up asterisk and two clients on the same network and it worked fine. I now have asterisk set up which is acting as a sip registrar. It is behind nat. I also have two clients which are behind nat on two separate networks. I can no longer register the clients. I have set 'nat=yes' in the client config but is there something else I must do for the asterisk sevrer itself?... I am only able to give you a link which might help you. Benjamin on Asterisk always writes good articles about nat (and other things as well). Here is one of them: http://lists.digium.com/pipermail/asterisk-users/2004-October/068275. html You probably should read the whole thread. Click the link thread on that page and search in the new page for postings with the the subject Almost there--Remote connection. -- Fabian Müller ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. - --- ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No busy-tone
The Busy show be at priority 102 (n+101). Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815 On Mon, 8 Nov 2004, Eric Wieling wrote: Nicklas Bondesson wrote: Just like this? It doesn't seem to work though. [wx3trunk-outgoing] include = internal-sip-callers exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,T) exten = _X.,101,Busy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message has been categorized as Legitimate by Bayesian Analyzer. If you do not agree, please click on the link below to train the Analyzer. http://216.162.162.39/bt/a.aspx?M=C:%5Csmtpmail%5CBayesTraining%5C2004-11-08%5C2cc9c71461074051a6775f6d7cfd9a8aC=2 -- --- This message has been inspected by DynaComm i:mail --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX TNT SIP / Asterisk
On Tue, 2 Nov 2004, James Taylor wrote: I can't get my MAX TNT to register with Asterisk. TAOS 11.0. SIP phone registeration show up in Asterisk like this: sip:[EMAIL PROTECTED] and works. The TNT shows up as: sip:@ip_address. Does anyone have this working? Am I missing something here? Where does the TNT get it's user name? Or, can it work without one? It works without one. Why do you need to register TNT to asterisk anyway? --alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX TNT
On Sun, 7 Nov 2004, voip wrote: Any body using Asterisk with a MAX TNT? SIP or H.323? asterisk + ser + TNT work fine. ser is proxy server, asterisk is feature server. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ZyXEL 2000w unregistering and no audio
Hi all, I'm trying to get a ZyXEL 2000w (with newest firmware) working with our in-office Asterisk 1.0 server. We have other SIP phones working. I've set up the Zyxel using the web interface and having it using g711ulaw compression. The first call after restarting the phone seems to work great. After that, however, things go down hill. The phone randomly starts saying Unregistered, and will still dial but will not play any audio, or send any audio (the line picking up a call from the 2000w doesn't hear anything). Is anyone else having this problem, or has anyone fixed this problem? Thanks, Chris -- Chris TenHarmsel Software Journeyman Atomic Object, LLC ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No busy-tone
Hi, I think you want this to be 102 since a busy returns n+101 n being the priority your Dial function was called. exten = _X.,101,Busy should be exten = _X.,102,Busy HTH -b Quoting Eric Wieling [EMAIL PROTECTED]: Nicklas Bondesson wrote: Just like this? It doesn't seem to work though. [wx3trunk-outgoing] include = internal-sip-callers exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,T) exten = _X.,101,Busy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Astricon Brazil. Why not ?! an Mexico too!!!!!!!!!!!!
I'll put Mexico in the row, I'll like to organize one here, of course with the help of the community and Digium. Regards Humberto On Mon, 08 Nov 2004 09:57:54 -0200, Rodrigo P. Telles [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jefferson, Jefferson Carvalho wrote: | Hello list , | | I'm looking for partners in Brazil to discuss a possible way | to have in Brazil an Official Conference regards Asterisk. Yes, it will be wonderfull ! | It'll includes a hardware/workshop and tech-seminars. | Would be nice if we could include in this conference , Anatel's | presence and a seminar about the lawful aspects of VoIP in Brazil. | I'm 100% sure that in Brazil , we have enough resources to | become a large and active Asterisk community. :) I'm glad to say that you are right and more, there are a lot of people in Brazil working with Asterisk. Contact me in private if you want. Best regards. | | Best Regards, | | -Jefferson Carvalho | Jeff Networks Consulting Ltda. | Teresina-PI-Brazil | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http://www.devel-it.com.br TDKOM Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBj19BiLK8unYgEMQRAqG4AJ0azCrspMj2Ca0m/bc6FERBf2lP6QCfYKEZ oguuMN/B5xP8WofrQppKI6Y= =9dRq -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 220 (or other phones) - line apperances?
Essentially I wish to have buttons on a panel (like the Snom 220's extension board) that show when people are on the phone or off the phone for a receptionist. As far as I know, you can't do this with asterisk, at least not easily. From what I've read, most people call this shared lines or something similar. I've heard that MGCP does support something similar to this, but that Asterisk does not specifically support it. Ahh, but you can with some phonesLook at the hint() Priority ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zap FXO channel locked up with steadystatic( white noise)
Title: RE: [Asterisk-Users] Zap FXO channel locked up with steadystatic(white noise) A similar problem just occurred to my Asterisk system last week for the first time. I had been running version 1.0 RC and then 1.0 release with no problem for the last two months. Then last week I upgraded to 1.0.2 and two days later we were doing an 8 party conference via PRI connection on a T100P and about 20 minutes into the conference I heard a short, soft beep and was disconnected (everyone was disconnected). Upon trying to call back into the meetme application it answered asked for the conf # and Pin, then connected to steady static. We ended up having to reboot the server to fix. I have since downgraded to 1.0 have not seen the problem occur again. Is there a known problem that would cause this in 1.0.2? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Richard Scobie Sent: Monday, November 08, 2004 12:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zap FXO channel locked up with steadystatic(white noise) Damon Estep wrote: I'm having the same problem on my TDM40B (FXS). Unloading and loading the modules seems to fix it temporally. Digium is sending me a replacement. Hopefully that will fix it. I plan to call tech support and see what they have to say, hopefully it is just defective and not un-reliable. Have you heard other complaints of the same thing? I have had the same issues with the FXO modules on systems that have run fine for a year using X100Ps. Rebooting the box has been required to fix it. Please let us know what Digium tell you. Regards, Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Errors
Okay, I found it.. I've never seen php installed in /usr/local/bin/php... But hey there's s first for everything eh? I'll remember this one for sure! Thanks everyone On Mon, 08 Nov 2004 07:27:03 -0800, Matthew Asham [EMAIL PROTECTED] wrote: As Steven asked, what about /usr/bin/php ? On Mon, 2004-11-08 at 05:39, Mike Roberts wrote: The Script http://pastebin.ca/1968 The File [EMAIL PROTECTED] agi-bin]# ll -rwxr-xr-x1 root root 1020 Nov 8 01:17 php-agi.agi [EMAIL PROTECTED] agi-bin]# pwd /var/lib/asterisk/agi-bin The Error *CLI -- Executing Answer(SIP/asterisk-6520, ) in new stack -- Executing AGI(SIP/asterisk-6520, php-agi.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/php-agi.agi Failed to execute '/var/lib/asterisk/agi-bin/php-agi.agi': No such file or directory -- AGI Script php-agi.agi completed, returning 0 -- Executing DigitTimeout(SIP/asterisk-6520, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(SIP/asterisk-6520, 15) in new stack -- Set Response Timeout to 15 -- Executing Read(SIP/asterisk-6520, Secret|IVR/en_enter_destination|0) in new stack -- Playing 'IVR/en_enter_destination' (language 'en') Extensions.conf [tf-did] exten = 877XXX,1,ANSWER exten = 877XXX,2,agi(php-agi.agi) exten = 877XXX,3,DigitTimeout,5 exten = 877XXX,4,ResponseTimeout,15 exten = 877XXX,5,Read(Secret,IVR/en_enter_destination,0) exten = 877XXX,6,dial(SIP/[EMAIL PROTECTED]) Hope that helps On Mon, 08 Nov 2004 05:16:24 -0600, Steven Critchfield [EMAIL PROTECTED] wrote: On Mon, 2004-11-08 at 03:00 -0800, Mike Roberts wrote: I'm having troubles with my agi scripts. -- Executing Answer(SIP/asterisk-7f82, ) in new stack -- Executing AGI(SIP/asterisk-7f82, php-agi.agi) in new stack Failed to execute '/var/lib/asterisk/agi-bin/php-agi.agi': No such file or directory Now that file is there! Thats a fact. The permissions are right (I hope) and I pulled the script off a working server. I had a cvs, I updated to 1.0 (where the script came from) and still nothing. Any ideas? Not unless you provide some more details. Why don't you paste in a ls -l of the above quoted file with full path? Also verify the she-bang line. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward incoming SIP calls to H323 ipphone?
Michael, Yeah.. for sure the channel is loaded.. calling to my asterisks works fine. I have included the oh323.conf and the original message. Thanks a lot for you help. I would would like to get this baby working. Alex The log; Nov 8 18:04:01 WARNING[294930]: channel.c:1901 ast_request: No channel type registered for 'OH323' Nov 8 18:04:01 NOTICE[294930]: app_dial.c:742 dial_exec: Unable to create channel of type 'OH323' Extensions.conf exten = 495234,3,Dial(OH323/192.168.1.20) oh323.conf; ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Configure TCP port range to be used by H.323 ; tcpStart=1 tcpEnd=2 ; ; Configure UDP port range to be used by H.323 ; Note: The port range used by RTP are configured from ; rtp.conf ; udpStart=1 udpEnd=2 ; ; Enable fast start (yes,no). ; fastStart=no ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=no ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=none ; ; Set the maximum number of inbound/outbound/simultaneous ; H.323 connections. ; outboundMax=10 inboundMax=10 simultaneousMax=10 ; ; Set the bandwidth limit for H.323 connections. ; The value is in Kbps. ; ;bandwidthLimit=1024 ; ; Set tracing options for the wrapper library and for the ; OpenH323 library. ; libTraceFile can be 'stdout' or a full path name to the tracefile. ; Only trace info for OpenH323 is logged in libTraceFile. ; wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; ;gatekeeper=192.168.1.2 gatekeeper=DISCOVER ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the gatekeeper registration timeout ; gatekeeperTTL=600 ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=TONE ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; context=voip-h323 ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=ASTERISK alias=666 ; ; Aliases/prefixes routed in more-aliases context. ; context=more-aliases alias=665 ; ; Aliases/prefixes routed in all-prefixes context. ; context=all-prefixes gwprefix=00 gwprefix=01 ; ; Aliases/prefixes routed in more-stuff context. ; context=more-stuff alias=664 gwprefix=02 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726- G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=G711A frames=20 ;codec=G711U ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=2 ;codec=G729 ;frames=2 On 8-nov-04, at 11:09, Michael Manousos wrote: Since you are able to receive H.323 calls with chan_oh323, I assume that the module is loaded. You could check the incoming/outgoing/simultaneous limits or submit the
[Asterisk-Users] Error forwarding calls to Voicemail from SER
Hello
I have to insist in this issue since I´ve done several test using Sems
and asterisk with very simple configuration files including the
original example from ser-cvs... in brief: if I call to a user who
belongs to voicemail group and I cancel the call before VM forward
routine begin then an invite is sent to a voicemail server
generating and sending a file with No audio, and I cant account this
call with Sip-Response-Code=487 (just an start record without
stop)...
does someone know how to solve this problem
thanks in advance
Rafael
PS:
ser.cfg and asterisk debug for this test:
#
# SER SIMPLE CFG for VM without acc...
# --- global configuration parameters
#debug=3 # debug level (cmd line: -dd)
#fork=yes
#log_stderror=no# (cmd line: -E)
#/* Uncomment these lines to enter debugging mode
debug=9
fork=yes
log_stderror=yes
#*/
listen=127.0.0.1
port=5060
# simple proxy script for forwarding to voicemail server
# for unavailable users
#
loadmodule /usr/local/lib/ser/modules/sl.so
loadmodule /usr/local/lib/ser/modules/tm.so
loadmodule /usr/local/lib/ser/modules/rr.so
loadmodule /usr/local/lib/ser/modules/maxfwd.so
loadmodule /usr/local/lib/ser/modules/mysql.so
loadmodule /usr/local/lib/ser/modules/group.so
loadmodule /usr/local/lib/ser/modules/usrloc.so
loadmodule /usr/local/lib/ser/modules/registrar.so
# time to give up on ringing -- global timer, applies to
#all transactions
modparam(tm, fr_inv_timer, 90)
# database with user group membership
modparam(group, db_url, mysql://ser:[EMAIL PROTECTED]/ser)
# - request routing logic ---
route {
if (!mf_process_maxfwd_header(10)) {
log(LOG: Too many hops\n);
sl_send_reply(483, Alas Too Many Hops);
break;
};
if (!(method==REGISTER)) record_route();
if (loose_route()) {
t_relay();
break;
};
if (!uri==myself) {
t_relay();
break;
};
if (method == REGISTER) {
if (!save(location)) {
sl_reply_error();
};
break;
};
# does the user wish redirection on no availability? (i.e., is he
# in the voicemail group?) -- determine it now and store it in
# flag 4, before we rewrite the flag using UsrLoc
if (is_user_in(Request-URI, voicemail)) {
setflag(4);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup(location)) {
# handle user which was not found
route(4);
break;
};
# if user is on-line and is in voicemail group, enable redirection
if (method == INVITE isflagset(4)) {
t_on_failure(1);
};
t_relay();
}
# - handling of unavailable user --
route[4] {
# non-Voip -- just send off-line
if (!(method == INVITE || method == ACK || method == CANCEL)) {
sl_send_reply(404, Not Found);
break;
};
# not voicemail subscriber
if (!isflagset(4)) {
sl_send_reply(404, Not Found and no voicemail turned on);
break;
};
# forward to voicemail now
rewritehostport(200.110.2.132:5060);
t_relay_to_udp(200.110.2.132, 5060);
}
# if forwarding downstream did not succeed, try voicemail running
# at 200.110.2.132:5060
failure_route[1] {
revert_uri();
rewritehostport(200.110.2.132:5060);
append_branch();
t_relay_to_udp(200.110.2.132, 5060);
}
Asterisk Voicemail server sip debug:
___-
*CLI
Sip read:
INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0
Record-Route: sip:200.110.2.131;ftag=bb0036aea4;lr=on
Via: SIP/2.0/UDP 200.110.2.131;branch=z9hG4bKe24b.b9e800b5.1
Via: SIP/2.0/UDP 10.0.1.27:5060;rport=5060;branch=z9hG4bKbb0036aea4125
From: sip:[EMAIL PROTECTED];tag=bb0036aea4
To: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 125 INVITE
Supported: timer, replaces
Min-SE: 1800
Date: Sun, 05 Jul 1970 12:53:15 GMT
User-Agent: AddPac SIP Gateway
Contact: sip:[EMAIL PROTECTED]:5060
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 285
Max-Forwards: 16
P-hint: usrloc applied
-- Forwarded message --
From: Rafael J. Risco G.V. [EMAIL PROTECTED]
Date: Fri, 5 Nov 2004 14:53:29 -0500
Subject: Voicemail: Strange behavior if caller-user cancels the call
To: [EMAIL PROTECTED], [EMAIL PROTECTED]
Hi
I need to solve this problem I´ve reported several times:
please take a look into this report:
http://mail.iptel.org/pipermail/serusers/2004-August/010930.html
...If the called-user belongs to the voicemail group and the
caller-user
FW: [Asterisk-Users] Need a creative solution - stop forwarding from changing caller ID
Nobody responded so Im sending this out again. I need help on stopping the Change caller ID on forward trick that either Cisco or Asterisk keeps doing. My upstream provider doesnt like it. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan Sent: Friday, November 05, 2004 3:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Need a creative solution - Caller ID and a stupidupstream Ok. Our upstream provider, IDS Telecom, will not let us set outbound caller ID to anything we want, like we used to be able to do with Expedius. We have to provide them with a list of any numbers we want to be able to set outbound caller ID to. So we have to give them a list of all our DIDs, from them, from Expedius, from VoicePulse, etc. etc. and its quite tedius, but weve been ok with it for now. However, we use Cisco 79xx phones. If we use the CFwdAll option on the phone, to forward calls, the phone will see the incoming call and redirect it to the call forwarded number, fine no problem. However, the phone (or asterisk) tries to change the outbound caller ID to the callers caller ID, this way the person receiving the call will know whos calling them. However, IDS will not recognize this number and wont let us set our caller ID to it so it will use the fall-back caller ID number, which is our companys main number. Just got a complaint from a customer whos upset that when he call forwards to his cell phone, whenever a call comes into his cell phone (relayed through their office phone), the caller ID shows our number, and not his office number or the callers number. If they picked up the phone, placed a call to their cell, the right caller Id would be provided, as I have this set as their caller id in sip.conf but only when the phone tries to do a blind transfer will it attempt to change or alter its normal outbound caller id, which seems to override whats in asterisks sip.conf file. Any ideas? All Ive got so far is: Call IDS and tell them to make any unknown outbound caller id numbers, just out of area. So when a call is forwarded through a Cisco, their cell will say Out of Area instead of our main number Disable the CFwdAll option on the Cisco phones (I dont know how though) Get Asterisk to ignore the phone changing the caller id information or override it If anybody can help me accomplish option 2 or 3, or has a better solution, it would be much appreciated. Life was easier when we used Expedius, they didnt care what we set the caller ID to. Too bad they lacked in too many other departments. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
FW: [Asterisk-Users] Need a creative solution - Caller ID and a stupidupstream
Nobody responded so Im sending this out again. I need help on stopping the Change caller ID on forward trick that either Cisco or Asterisk keeps doing. My upstream provider doesnt like it. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan Sent: Friday, November 05, 2004 3:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Need a creative solution - Caller ID and a stupidupstream Ok. Our upstream provider, IDS Telecom, will not let us set outbound caller ID to anything we want, like we used to be able to do with Expedius. We have to provide them with a list of any numbers we want to be able to set outbound caller ID to. So we have to give them a list of all our DIDs, from them, from Expedius, from VoicePulse, etc. etc. and its quite tedius, but weve been ok with it for now. However, we use Cisco 79xx phones. If we use the CFwdAll option on the phone, to forward calls, the phone will see the incoming call and redirect it to the call forwarded number, fine no problem. However, the phone (or asterisk) tries to change the outbound caller ID to the callers caller ID, this way the person receiving the call will know whos calling them. However, IDS will not recognize this number and wont let us set our caller ID to it so it will use the fall-back caller ID number, which is our companys main number. Just got a complaint from a customer whos upset that when he call forwards to his cell phone, whenever a call comes into his cell phone (relayed through their office phone), the caller ID shows our number, and not his office number or the callers number. If they picked up the phone, placed a call to their cell, the right caller Id would be provided, as I have this set as their caller id in sip.conf but only when the phone tries to do a blind transfer will it attempt to change or alter its normal outbound caller id, which seems to override whats in asterisks sip.conf file. Any ideas? All Ive got so far is: Call IDS and tell them to make any unknown outbound caller id numbers, just out of area. So when a call is forwarded through a Cisco, their cell will say Out of Area instead of our main number Disable the CFwdAll option on the Cisco phones (I dont know how though) Get Asterisk to ignore the phone changing the caller id information or override it If anybody can help me accomplish option 2 or 3, or has a better solution, it would be much appreciated. Life was easier when we used Expedius, they didnt care what we set the caller ID to. Too bad they lacked in too many other departments. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 220 (or other phones) - line
Essentially I wish to have buttons on a panel (like the Snom 220's extension board) that show when people are on the phone or off the phone for a receptionist. As far as I know, you can't do this with asterisk, at least not easily. From what I've read, most people call this shared lines or something similar. I've heard that MGCP does support something similar to this, but that Asterisk does not specifically support it. Ahh, but you can with some phonesLook at the hint() Priority Hey Thanks for the info! Wow, that makes the Snom 190/200 perfect for a small office of 2-6 users. You can switch them away from their old crappy PBX, and you don't even have to retrain them! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Setting DND feature via access code
I commonly use the DND feature in Asterisk by dialing *78. When I do this I hear about a second of stutter dialtone to let me know the feature was set. Is it possible to configure the Zap channel to continue to provide stutter dialtone while the line is in DND? This way if someone forgets to turn it off or is unaware that it is even on, they would have some indication as soon as they go off-hook. Thanks, --LJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward incoming SIP calls to H323 ipphone?
Alex van Es wrote: Michael, Yeah.. for sure the channel is loaded.. calling to my asterisks works fine. I have included the oh323.conf and the original message. Thanks a lot for you help. I would would like to get this baby working. Alex The log; Nov 8 18:04:01 WARNING[294930]: channel.c:1901 ast_request: No channel type registered for 'OH323' Hmm, according to this message, chan_oh323.so isn't loaded. Your config is fine. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [Asterisk-Users] Need a creative solution - Caller ID and a stupidupstream
Paul Rodan wrote: Nobody responded so Im sending this out again. I need help on stopping the Change caller ID on forward trick that either Cisco or Asterisk keeps doing. My upstream provider doesnt like it. This doesn't help? 'f' -- Forces callerid to be set as the extension of the line making/redirecting the outgoing call. For example, some PSTNs don't allow callerids from other extensions then the ones that are assigned to you. Of course you have to do a show application dial to see all the Dial() options. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Limit DTMF Tones
Ok I figured it out mostly, I went by what Flynn posted. I commented out the particular lines in res_features.c when a call is connected no DTMF is passed. The only problem I am having now is Im not sure how to set up the IVR. This is what I need to be done. Sounds simple and probably is, but I have never used the IVR function in *. I would like the following to happen for a person to be able to dial out. --- The person goes to phone dials 2 --- IVR answers and speaks Please enter the destination phone number --- The persons enters the phone number they would like to dial --- IVR Then says Enter your 6 digit PIN number --- The person enters their PIN number --- * Then dials the phone number that was entered, pauses 5 seconds then dials the PIN number that was entered. * does not need to process these numbers in any way, It does not need to check PIN numbers. I just want * to send these numbers. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Macro issue.
Hi, Anyone know how to get voicemail to continue running the next exten in the dialplan when a user hangs up. If a user hits # after leaving a message instead of hanging, up it works. I am trying to do a call back macro and when users hangup after leaving a voicemail the rest of my macro does not run. Any help would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward incoming SIP calls to H323 ipphone?
Michael, When I do show modules it shows up in the list.. And if it wasn't loaded, how come asterisks can still receive h323 calls? Alex apeldoorn*CLI> show modules ModuleDescription Use Count chan_modem.so Generic Voice Modem Driver 0 chan_modem_aopen.so A/Open (Rockwell Chipset) ITU-2 VoiceMod 0 res_musiconhold.soMusic On Hold Resource 1 res_adsi.so ADSI Resource1 res_features.so Call Parking Resource1 res_crypto.so Cryptographic Digital Signatures 1 res_indications.soIndications Configuration0 res_monitor.soCall Monitoring Resource 1 res_agi.soAsterisk Gateway Interface (AGI) 0 chan_sip.so Session Initiation Protocol (SIP)0 chan_modem_bestdata.soBestData (Conexant V.90 Chipset) VoiceMo 0 chan_modem_i4l.so ISDN4Linux Emulated Modem Driver 0 chan_agent.so Agent Proxy Channel 0 chan_mgcp.so Media Gateway Control Protocol (MGCP)0 chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 chan_local.so Local Proxy Channel 0 chan_skinny.soSkinny Client Control Protocol (Skinny) 0 chan_oss.so OSS Console Channel Driver 0 chan_phone.so Linux Telephony API Support 0 pbx_config.so Text Extension Configuration 0 pbx_wilcalu.soWil Cal U (Auto Dialer) 0 pbx_spool.so Outgoing Spool Support 1 app_dial.so Dialing Application 0 app_playback.so Trivial Playback Application 0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_directory.so Extension Directory 0 app_mp3.soSilly MP3 Application0 app_system.so Generic System() application 0 app_echo.so Simple Echo Application 0 app_record.so Trivial Record Application 0 app_image.so Image Transmission Application 0 app_url.soSend URL Applications0 app_disa.so DISA (Direct Inward System Access) Appli 0 app_qcall.so Call from Queue 0 app_adsiprog.so Asterisk ADSI Programming Application0 app_getcpeid.so Get ADSI CPE ID 0 app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 app_zapateller.so Block Telemarketers with Special Informa 0 app_setcallerid.soSet CallerID Application 0 app_festival.so Simple Festival Interface0 app_queue.so True Call Queueing 0 app_senddtmf.so Send DTMF digits Application 0 app_parkandannounce.soCall Parking and Announce Application0 app_striplsd.so Strip trailing digits0 app_setcidname.so Set CallerID Name0 app_lookupcidname.so Look up CallerID Name from local databas 0 app_substring.so (Deprecated) Save substring digits in a 0 app_macro.so Extension Macros 0 app_authenticate.so Authentication Application 0 app_softhangup.so Hangs up the requested channel 0 app_lookupblacklist.soLook up Caller*ID name/number from black 0 app_waitforring.soWaits until first ring after time0 app_privacy.soRequire phone number to be entered, if n 0 app_db.so Database access functions for Asterisk e 0 app_chanisavail.soCheck if channel is available0 app_enumlookup.so ENUM Lookup 0 app_transfer.so Transfer 0 app_setcidnum.so Set CallerID Number 0 app_cdr.soMake sure asterisk doesn't save CDR for 0 app_hasnewvoicemail.soIndicator for whether a voice mailbox ha 0 app_sayunixtime.soSay time 0
[Asterisk-Users] Sort of OT: Grandstream Phone and MS Wireless mouse
I have a Grandstream 101 phone on my desk. I also use a Microsoft Wireless Optical mouse. When I'm using the phone, the mouse doesn't work very well - herky jerky movement. If I move the phone away from the mouse and receiver, they work fine again - otherwise I need them very close together to get the mouse to work. Has anyone else experienced this? I assume it's not fixable other than move the phone away from the mouse? Do all phones have this problem or if I upgrade to a better quality phone, it'll work fine? Makarios Communications, LLC Network Monitoring, Consulting, Web Hosting www.makarios.us [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward incoming SIP calls to H323 ipphone?
What to answer to this one? Module loaded and no 'OH323' channel type registered? How did you do that? As a last attempt, enable debugging on the console (logger.conf) and start Asterisk with -vvvcd, rerun and email the full output. Also, send the portion of Asterisk boot messages (where it loads the various modules) that belong to chan_oh323.so. Michael. Alex van Es wrote: Michael, When I do show modules it shows up in the list.. And if it wasn't loaded, how come asterisks can still receive h323 calls? Alex apeldoorn*CLI show modules Module Description Use Count chan_modem.so Generic Voice Modem Driver 0 chan_modem_aopen.so A/Open (Rockwell Chipset) ITU-2 VoiceMod 0 res_musiconhold.so Music On Hold Resource 1 res_adsi.so ADSI Resource 1 res_features.so Call Parking Resource 1 res_crypto.so Cryptographic Digital Signatures 1 res_indications.so Indications Configuration 0 res_monitor.so Call Monitoring Resource 1 res_agi.so Asterisk Gateway Interface (AGI) 0 chan_sip.so Session Initiation Protocol (SIP) 0 chan_modem_bestdata.so BestData (Conexant V.90 Chipset) VoiceMo 0 chan_modem_i4l.so ISDN4Linux Emulated Modem Driver 0 chan_agent.so Agent Proxy Channel 0 chan_mgcp.so Media Gateway Control Protocol (MGCP) 0 chan_iax2.so Inter Asterisk eXchange (Ver 2) 0 chan_local.so Local Proxy Channel 0 chan_skinny.so Skinny Client Control Protocol (Skinny) 0 chan_oss.so OSS Console Channel Driver 0 chan_phone.so Linux Telephony API Support 0 pbx_config.so Text Extension Configuration 0 pbx_wilcalu.so Wil Cal U (Auto Dialer) 0 pbx_spool.so Outgoing Spool Support 1 app_dial.so Dialing Application 0 app_playback.so Trivial Playback Application 0 app_voicemail.so Comedian Mail (Voicemail System) 0 app_directory.so Extension Directory 0 app_mp3.so Silly MP3 Application 0 app_system.so Generic System() application 0 app_echo.so Simple Echo Application 0 app_record.so Trivial Record Application 0 app_image.so Image Transmission Application 0 app_url.so Send URL Applications 0 app_disa.so DISA (Direct Inward System Access) Appli 0 app_qcall.so Call from Queue 0 app_adsiprog.so Asterisk ADSI Programming Application 0 app_getcpeid.so Get ADSI CPE ID 0 app_milliwatt.so Digital Milliwatt (mu-law) Test Applicat 0 app_zapateller.so Block Telemarketers with Special Informa 0 app_setcallerid.so Set CallerID Application 0 app_festival.so Simple Festival Interface 0 app_queue.so True Call Queueing 0 app_senddtmf.so Send DTMF digits Application 0 app_parkandannounce.so Call Parking and Announce Application 0 app_striplsd.so Strip trailing digits 0 app_setcidname.so Set CallerID Name 0 app_lookupcidname.so Look up CallerID Name from local databas 0 app_substring.so (Deprecated) Save substring digits in a 0 app_macro.so Extension Macros 0 app_authenticate.so Authentication Application 0 app_softhangup.so Hangs up the requested channel 0 app_lookupblacklist.so Look up Caller*ID name/number from black 0 app_waitforring.so Waits until first ring after time 0 app_privacy.so Require phone number to be entered, if n 0 app_db.so Database access functions for Asterisk e 0 app_chanisavail.so Check if channel is available 0 app_enumlookup.so ENUM Lookup 0 app_transfer.so Transfer 0 app_setcidnum.so Set CallerID Number 0 app_cdr.so Make sure asterisk doesn't save CDR for 0 app_hasnewvoicemail.so Indicator for whether a voice mailbox ha 0 app_sayunixtime.so Say time 0 app_cut.so Cuts up variables 0 app_read.so Read Variable Application 0 app_setcdruserfield.so CDR user field apps 0 app_random.so Random goto 0 app_ices.so Encode and Stream via icecast and ices 0 app_eval.so Reevaluates strings 0 app_nbscat.so Silly NBS Stream Application 0 app_sendtext.so Send Text Applications 0 app_exec.so Executes applications 0 app_sms.so SMS/PSTN handler 0 app_groupcount.so Group Management Routines 0 app_txtcidname.so TXTCIDName 0 app_controlplayback.so Control Playback Application 0 app_talkdetect.so Playback with Talk Detection 0 app_alarmreceiver.so Alarm Receiver for Asterisk 0 app_userevent.so Custom User Event Application 0 app_verbose.so Send verbose output 0 app_test.so Interface Test Application 0 app_forkcdr.so Fork The CDR into 2 seperate entities. 0 codec_ilbc.so iLBC/PCM16 (signed linear) Codec Transla 0 codec_gsm.so GSM/PCM16 (signed linear) Codec Translat 0 codec_lpc10.so LPC10 2.4kbps (signed linear) Voice Code 0 codec_adpcm.so Adaptive Differential PCM Coder/Decoder 0 codec_ulaw.so Mu-law Coder/Decoder 0 codec_alaw.so A-law Coder/Decoder 0 codec_a_mu.so A-law and Mulaw direct Coder/Decoder 0 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_gsm.so Raw GSM data 0 format_wav.so Microsoft WAV format (8000hz Signed Line 0 format_wav_gsm.so Microsoft WAV format (Proprietary GSM) 0 format_vox.so Dialogic VOX (ADPCM) File Format 0 format_pcm.so Raw uLaw 8khz Audio support (PCM) 0 format_g729.so Raw G729 data 0 format_pcm_alaw.so Raw aLaw 8khz PCM Audio support 0 format_h263.so Raw h263 data 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0
RE: FW: [Asterisk-Users] Need a creative solution - Caller ID anda stupidupstream
Hmmm... You're right, I must have missed that option. If this works, I do
apologize for wasting your valuable time. However, do I put this on the
outbound or inbound rule?
This rule:
; Local
exten = _9NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
exten = _9NXXNXX,2,Congestion
or do I put this on the actual extension of the person who has the CFwdALL
option set, this rule:
exten = 3024,1,Dial(SIP/sales1,20,r)
exten = 3024,2,VoiceMail(u3043)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Monday, November 08, 2004 12:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [Asterisk-Users] Need a creative solution - Caller ID anda
stupidupstream
Paul Rodan wrote:
Nobody responded so I'm sending this out again. I need help on stopping
the Change caller ID on forward trick that either Cisco or Asterisk
keeps doing. My upstream provider doesn't like it.
This doesn't help?
'f' -- Forces callerid to be set as the extension of the line
making/redirecting the outgoing call. For example, some PSTNs
don't allow callerids from other extensions then the ones
that are assigned to you.
Of course you have to do a show application dial to see all the Dial()
options.
___
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___
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[Asterisk-Users] timing and dropped calls
Hi, I have a * server which does only SIP to H323 completely in IP domain, there is no digium h/w in it. In your experience, for this type of application, is it required to have a timing source toprevent the calls being dropped. Cheers SW ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Forward incoming SIP calls to H323 ipphone?
Michael,
Attached some of the logging.
I noticed that when I call the sip number, it surely is talking to my
ipphone. When I look at the debug info coming out of my
phone it starts to spit out information (not readable) so for sure
asterisk and the phone are talking.
I tried setting a different codec in the oh323.conf, but that didn't
help..
Alex
Asterisk Ready.
*CLI -- H.323 call to 192.168.1.20 with codec ALAW
Urgent handler
-- Called 192.168.1.20
Urgent handler
-- H.323 call 'ip$localhost/24187' cleared, reason 22 (Remote
endpoint is offline)
-- Hungup 'OH323/L24187'
== No one is available to answer at this time
Urgent handler
Asterisk Dynamic Loader Starting:
[chan_oh323.so] = (OpenH323 Channel Driver)
[1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323
v1.13.5, PWlib v1.6.6
== OpenH323 Channel Ready (v0.6.3)
Nov 8 19:28:16 DEBUG[147465]: build_route: Record-Route hop:
sip:[EMAIL PROTECTED];ftag=999500031330;lr=on
Nov 8 19:28:16 DEBUG[147465]: build_route: Contact hop:
sip:82.161.62.10:5060
Nov 8 19:28:16 DEBUG[294930]: Launching 'System'
Nov 8 19:28:16 DEBUG[294930]: Launching 'System'
Nov 8 19:28:16 DEBUG[294930]: Launching 'Dial'
Nov 8 19:28:16 DEBUG[294930]: In oh323_request: type=OH323, format=8,
data=192.168.1.20.
Nov 8 19:28:16 DEBUG[294930]: Player fds 27,28 - Recorder fds 29,30 -
Event pipe 31,40.
Nov 8 19:28:16 DEBUG[294930]: Created new call structure 0 (5548
bytes).
Nov 8 19:28:16 DEBUG[294930]: OH323/L0: Raw format set to ALAW.
Nov 8 19:28:16 DEBUG[294930]: Context is 'voip-h323', extension is 's'.
Nov 8 19:28:16 DEBUG[294930]: CallerID/ANI is ''.
Nov 8 19:28:16 DEBUG[294930]: OH323/L0: Native format changed to ALAW.
Nov 8 19:28:16 DEBUG[294930]: In oh323_call (OH323/L0,
dest=192.168.1.20, timeout=0).
Nov 8 19:28:16 DEBUG[294930]: OH323/L0: Generating CallerID 'Alex
82.161.62.10'
Nov 8 19:28:16 DEBUG[294930]: CID is '82.161.62.10'.
Nov 8 19:28:16 DEBUG[294930]: CIDname is 'Alex'.
Nov 8 19:28:16 DEBUG[294930]: OH323/192.168.1.20: No ${OH323_OUTCODEC}.
Nov 8 19:28:16 DEBUG[294930]: capability_set[0] - 2
Nov 8 19:28:16 DEBUG[294930]: capability_set[1] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[2] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[3] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[4] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[5] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[6] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[7] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[8] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[9] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[10] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[11] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[12] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[13] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[14] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[15] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[16] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[17] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[18] - 0
Nov 8 19:28:16 DEBUG[294930]: capability_set[19] - 0
Nov 8 19:28:16 DEBUG[294930]: NEW STATE: NULL -- INIT
Nov 8 19:28:16 DEBUG[294930]: OH323/L24187: Call to 192.168.1.20
initiated successfully.
Nov 8 19:28:16 DEBUG[294930]: Set channel OH323/L24187 to read format
ULAW
Nov 8 19:28:16 DEBUG[294930]: Set channel SIP/5-785e to write
format ULAW
Nov 8 19:28:16 DEBUG[294930]: Set channel OH323/L24187 to write format
ALAW
Nov 8 19:28:16 DEBUG[294930]: Set channel SIP/5-785e to read
format ALAW
Nov 8 19:28:26 DEBUG[49155]: ENTER cleanup_h323_connection.
Nov 8 19:28:26 DEBUG[49155]: Call ip$localhost/24187 found in 0.
Nov 8 19:28:26 DEBUG[49155]: Call ip$localhost/24187 cleared in INIT
state.
Nov 8 19:28:26 DEBUG[49155]: NEW STATE: INIT -- CLEARED
Nov 8 19:28:26 DEBUG[49155]: Forcing H.323 channel to hangup.
Nov 8 19:28:26 DEBUG[294930]: OH323/L24187: Channel was shut down.
Nov 8 19:28:26 DEBUG[294930]: Hanging up channel 'OH323/L24187'
Nov 8 19:28:26 DEBUG[294930]: In oh323_hangup (OH323/L24187).
Nov 8 19:28:26 DEBUG[294930]: NEW STATE: CLEARED -- CLEARED
Nov 8 19:28:26 DEBUG[294930]: OH323/L24187: Call ip$localhost/24187
found in 0.
Nov 8 19:28:26 DEBUG[294930]: Releasing resources of call (0).
Nov 8 19:28:26 DEBUG[294930]: Releasing allocated resources (0).
Nov 8 19:28:26 DEBUG[294930]: Player fds 27,28 - Recorder fds 29,30 -
Event pipe 31,40.
Nov 8 19:28:26 DEBUG[294930]: Closing socket 28.
On 8-nov-04, at 18:53, Michael Manousos wrote:
What to answer to this one?
Module loaded and no 'OH323' channel type registered?
How did you do that?
As a last attempt, enable debugging on the console (logger.conf)
and start Asterisk with -vvvcd, rerun and email the full output.
Also, send the portion of Asterisk boot messages (where it loads
the various modules) that belong to chan_oh323.so.
Michael.
Alex van Es wrote:
Michael,
When I
Re: [Asterisk-Users] Cisco 7910 - Success?
I have two 7910's one is a 7910G+SW and one is 7910+SW
I have the 7910G+SW to work with an xml file in the /tftpboot directory.
Using chan_skinniny however I cannot get the hold tranfer etc. buttons to
work.
skinny.conf is as below:
---
501]
context=default
nat=no
host=192.168.10.144
accountcode=501
fromuser=501
callerid=Jim Forte 501
incominglimit=1
outgoinglimit=1
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=gsm
device=SEP000AF4A3D50A
version=P002F202
linelabel=JPF 501
callwaiting=yes
transfer=yes
threewaycalling=yes
line = 501
/tftboot/SEP000AF4A3D50A.cnf.xml file is as below:
-
device
devicePool
callManagerGroup
members
member priority=0
callManager
ports
ethernetPhonePort2000/ethernetPhonePort
/ports
processNodeName192.168.10.89/processNodeName
/callManager
/member
/members
/callManagerGroup
/devicePool
versionStamp{Jan 01 2002 00:00:00}/versionStamp
loadInformation/loadInformation
userLocale
nameEnglish_United_States/name
langCodeen/langCode
/userLocale
networkLocaleUnited_States/networkLocale
idleTimeout0/idleTimeout
authenticationURL/authenticationURL
directoryURL/directoryURL
idleURL/idleURL
informationURL/informationURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURL/servicesURL
loadInformation6 model=IP Phone 7910P00403020214/loadInformation6
/device
(END)
Any thoughts on how to get the hold and tranfer button working
appreciated.
jim forte
On Thu, 4 Nov 2004, Matthew Boehm wrote:
I know that the 7910 only works with Skinny. We have a possible client that
wants to bring 80 lines to us off his current provider. All 80 of his phones
are Cisco 7910s. Does anyone run 7910s with chan_sccp or the like and find
that it works good?
Thanks,
Matthew
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Yours Truly
James Forte, Magna.Net Inc. THE Dot Net in Timeshare
http://Timeshare.Magna.Net/ mailto:[EMAIL PROTECTED]
7540 Municipal Drive, Orlando FL, 407-352-2402 EFax: 253-423-5482
THIS COMMUNICATION IS ONLY INTENDED FOR THE RECIPIENT(S) ABOVE.
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[Asterisk-Users] Re: getting callerid from spa3k to asterisk
You could maybe look at the autocreatepeer option for sip.conf that level of vulnerability would not seem to be a good approach to solving some sort of sip/config problem :-) the problem is in the sip handshake between the spa3k and *. i have been hoping a sip geek would have a chance to look at it. randy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX TNT SIP / Asterisk
Your question indicates that there may be a better way... ??? I want to use the voice mail and extension features of Asterisk, and sometimes there is this NAT problem that Asterisk seems to handle very well. I've been using H.323 with the TNT. Do you have an alternate solution? On Mon, 8 Nov 2004 10:41:31 -0500 (EST), [EMAIL PROTECTED] wrote: On Tue, 2 Nov 2004, James Taylor wrote: I can't get my MAX TNT to register with Asterisk. TAOS 11.0. SIP phone registeration show up in Asterisk like this: sip:[EMAIL PROTECTED] and works. The TNT shows up as: sip:@ip_address. Does anyone have this working? Am I missing something here? Where does the TNT get it's user name? Or, can it work without one? It works without one. Why do you need to register TNT to asterisk anyway? --alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor 3505 Summerhll Road Suite 11 Texarkana, Texas 75503 903-793-1953 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Astricon Brazil. Why not ?!
Hi guys, I am so happy to hear that. I'm agree with Jefferson There a lot of people working with * here in Brazil. I would be happy to help you. I have a idea, what do you think about have a chat some day in this week with the Brazilian guys? Geraldo Santo Osasco-SP-Brazil -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Rodrigo P. Telles Enviada em: segunda-feira, 8 de novembro de 2004 09:58 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Astricon Brazil. Why not ?! -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jefferson, Jefferson Carvalho wrote: | Hello list , | | I'm looking for partners in Brazil to discuss a possible way | to have in Brazil an Official Conference regards Asterisk. Yes, it will be wonderfull ! | It'll includes a hardware/workshop and tech-seminars. | Would be nice if we could include in this conference , Anatel's | presence and a seminar about the lawful aspects of VoIP in Brazil. | I'm 100% sure that in Brazil , we have enough resources to | become a large and active Asterisk community. :) I'm glad to say that you are right and more, there are a lot of people in Brazil working with Asterisk. Contact me in private if you want. Best regards. | | Best Regards, | | -Jefferson Carvalho | Jeff Networks Consulting Ltda. | Teresina-PI-Brazil | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http://www.devel-it.com.br TDKOM Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBj19BiLK8unYgEMQRAqG4AJ0azCrspMj2Ca0m/bc6FERBf2lP6QCfYKEZ oguuMN/B5xP8WofrQppKI6Y= =9dRq -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 500 software?
Rich Adamson wrote: Polycom ships out two different phones , ones with H323,and one with SIP already loaded. Thank you, Steve Maroney Correction, the polycom IP 500 ships without h.323 or SIP software (it only has a bootrom on it), and software is only distributed by polycom authorized VoIP partners. I have personally taken issue with this as they advertise the product as H.323 and SIP compliant, yet without additional software it does not even know what SIP or H.323 are. That's not right. New phones come loaded with the current relevant firmware. Upgraded f/w is only available to/from certified resellers. Or look on the wiki for where it is freely available. The two new 500's that were purchased from a Polycom reseller actually came with no firmware installed at all; only the bootloader (or whatever its called). Someone on this list pointed me to a souce for downloading the sip image, and now I've got the phone running, but it won't register with *. Not sure what the registration problem is as yet, but doing a sip debug indicates the registration failure. I double checked the Auth UserID and Password and they appear to be correct. Seems others on the list have had the same issue, but I've not found any responses resolving the problem as yet. Anyone have any suggestions? Rich Following works for me (I have a IP600) At web interface (Registration1): Display Name: Rich Adamson Address: 1234 ;your extension Auth User ID: 1234 Auth Password: 1234 Label: 1234 At sip.conf [1234] type=friend username=1234 secret=1234 Hope this helps Jorge Mendoza ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail Macro issue.
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Monday, November 08, 2004 10:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Voicemail Macro issue. Hi, Anyone know how to get voicemail to continue running the next exten in the dialplan when a user hangs up. If a user hits # after leaving a message instead of hanging, up it works. I am trying to do a call back macro and when users hangup after leaving a voicemail the rest of my macro does not run. Any help would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This might work for you http://www.voip-info.org/wiki-Asterisk+tips+callback It will not work for us, because we need a repeated call out until the message is picked up, so I have posted a bounty, if the feature I am looking for interests you enough to contribute please add your contribution to the bounty, at some point it will be attractive enough for a coder to do the work. Details on the bounty are here http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20outcall %20notification%20application ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7910 - Success?
Have you tried chan_sccp?
http://chan-sccp.sourceforge.net
Matthew
- Original Message -
From: James Forte [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Monday, November 08, 2004 12:20 PM
Subject: Re: [Asterisk-Users] Cisco 7910 - Success?
I have two 7910's one is a 7910G+SW and one is 7910+SW
I have the 7910G+SW to work with an xml file in the /tftpboot directory.
Using chan_skinniny however I cannot get the hold tranfer etc. buttons to
work.
skinny.conf is as below:
---
501]
context=default
nat=no
host=192.168.10.144
accountcode=501
fromuser=501
callerid=Jim Forte 501
incominglimit=1
outgoinglimit=1
disallow=all
allow=ulaw
allow=alaw
allow=g723.1
allow=gsm
device=SEP000AF4A3D50A
version=P002F202
linelabel=JPF 501
callwaiting=yes
transfer=yes
threewaycalling=yes
line = 501
/tftboot/SEP000AF4A3D50A.cnf.xml file is as below:
-
device
devicePool
callManagerGroup
members
member priority=0
callManager
ports
ethernetPhonePort2000/ethernetPhonePort
/ports
processNodeName192.168.10.89/processNodeName
/callManager
/member
/members
/callManagerGroup
/devicePool
versionStamp{Jan 01 2002 00:00:00}/versionStamp
loadInformation/loadInformation
userLocale
nameEnglish_United_States/name
langCodeen/langCode
/userLocale
networkLocaleUnited_States/networkLocale
idleTimeout0/idleTimeout
authenticationURL/authenticationURL
directoryURL/directoryURL
idleURL/idleURL
informationURL/informationURL
messagesURL/messagesURL
proxyServerURL/proxyServerURL
servicesURL/servicesURL
loadInformation6 model=IP Phone 7910P00403020214/loadInformation6
/device
(END)
Any thoughts on how to get the hold and tranfer button working
appreciated.
jim forte
On Thu, 4 Nov 2004, Matthew Boehm wrote:
I know that the 7910 only works with Skinny. We have a possible client
that
wants to bring 80 lines to us off his current provider. All 80 of his
phones
are Cisco 7910s. Does anyone run 7910s with chan_sccp or the like and
find
that it works good?
Thanks,
Matthew
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--
--
Yours Truly
James Forte, Magna.Net Inc. THE Dot Net in Timeshare
http://Timeshare.Magna.Net/ mailto:[EMAIL PROTECTED]
7540 Municipal Drive, Orlando FL, 407-352-2402 EFax: 253-423-5482
THIS COMMUNICATION IS ONLY INTENDED FOR THE RECIPIENT(S) ABOVE.
PLEASE DISCARD IF YOU HAVE RECEIVED THIS IN ERROR.
-
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RE: [Asterisk-Users] New-B-ish Question
Thanks for your efforts Steve, but it turned out to be a problem with SJphone. X-Lite does not exhibit the same symptoms. Peter From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Friday, November 05, 2004 7:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] New-B-ish Question more info please. conf files and console output. - Original Message - From: Peter Awad To: [EMAIL PROTECTED] Sent: Friday, November 05, 2004 6:49 AM Subject: [Asterisk-Users] New-B-ish Question Ive been exploring asterisk for about 1 week now and have a server setup with 2 soft phones and an FXO. I can call between softphones, I can call into the PBX via FXO and route call to a Softphone. But when I call out the receiving phone rings once then the call terminates. Please tell me Im missing something obvious. Any help would be appreciated. Peter ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P card on Mac dialtone problem
I'm having trouble with a TDM400P card configured with one fxo and one fxs device. The system is a Mac G3 B/W running YellowDog 3.01 (2.4.22-2f kernel). The card is installed with a power cable, it configures itself properly at boot time, ztcfg and zttool shows everything is fine. Asterisk 1-0-28 starts fine with no errors and can sense a telephone handset going on and off hook on the fxs port. The simple switch starts up and stops when the handset goes back on-hook. The phone will also ring when a call is placed to it. On the fxo port it can sense an incoming call from the analog phone line, answer the call and detect hangup. However, neither the phone or the incoming call can receive or transmit anything. There's no dialtone when the phone is picked up, digit press are ignored and there is no sound heard on the incoming phone line when Asterisk playsback a sound file and cannot sense digit presses. ztmonitor shows that the fxs channel sends dialtone to the phone goes off-hook but nothing is received when a key is pressed on the phone. I've played with txgain and rxgain without any benefit. Anyone have any idea why ztmonitor would show that sound is being transmitted but yet there would still be no dialtone? Thanks, Stephen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No busy-tone
Thanks, it is finally working. Where can I find more info on the priorities in Asterisk? Nicklas -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bill Hamel Sent: den 8 november 2004 16:56 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No busy-tone Hi, I think you want this to be 102 since a busy returns n+101 n being the priority your Dial function was called. exten = _X.,101,Busy should be exten = _X.,102,Busy HTH -b Quoting Eric Wieling [EMAIL PROTECTED]: Nicklas Bondesson wrote: Just like this? It doesn't seem to work though. [wx3trunk-outgoing] include = internal-sip-callers exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,T) exten = _X.,101,Busy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by The CCIS.net MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. -- This message was sent using IMP, the Internet Messaging Program. -- This message has been scanned for viruses and dangerous content by the Bugs.Hamel.Net MailScanner, and appears to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Old Building Needs a New Telephone System
First, I will admit that I have not worked with PoE before so I'm asking this for my own benifit as well as the OP's benifit. Doesn't PoE require at lest 3 pairs to be availible? I know that pins 1, 2, 3, and 6 get used for ethernet communications and doesn't the power get transmitted over pins 4 and 5? Tim Donahue On Mon, 2004-11-08 at 00:00, Edward Beheler wrote: According to the spec sheet, they will do passthru PoE on the first jack. Ed Beheler On Sun, 07 Nov 2004 21:00:25 -0700, Michael Welter [EMAIL PROTECTED] wrote: Joe Greco wrote: We have a 100 year old building here in Colorado that needs a new Your best bet may be something like this: http://www.3com.com/products/en_US/detail.jsp?tab=featurespathtype=purchasesku=WEBBNCNJ220SYS I can't find a schematic for the IntelliJack--can I have Ethernet and PoE over two pair? -- Michael Welter Introspect Telephony Corp. Denver, Colorado US +1.303.674.2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Old Building Needs a New Telephone System
Tim Donahue wrote: First, I will admit that I have not worked with PoE before so I'm asking this for my own benifit as well as the OP's benifit. Doesn't PoE require at lest 3 pairs to be availible? I know that pins 1, 2, 3, and 6 get used for ethernet communications and doesn't the power get transmitted over pins 4 and 5? A PoE-enabled connection needs all four pairs. Two pairs for Tx/Rx, one pair for power, one pair for ground. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cordless vs Wireless phones
We currently have an Asterisk installation and need to add cordless / wireless phones. Requirements are these phone need to be equals to the wired devices, i.e. dedicated buttons for hold, transfer, etc. , e.g. not an ATA connected analog phone cordless phone. Was thinking of using 802.11b SIP phones (etc), but this opens up all the security concerns of 802.11 and the network. Do any of these phone support VPNs? Have to isolate the WLAN from the LAN. If not is there a SIP (or any other Asterisk channel) device that is a cordless phone. Some things like combining an ATA w/a cordless phone? But as one device with all the digital features? Thanks! Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] Old Building Needs a New Telephone System
My standard answer to POE questionsMostly stolen from or repeated in a Network Computing issue about 2 months ago. PoE factoids: PoE uses the spare pairs *or* the data pairs (which one to use is automatically detected) in an ethernet (10 or 10/100) cable to carry -48V dc from the power sourcing equipment (PSE) in an endpoint switch (or midspan hub) to the powered device (PD) appliance at the other end of the cable. Clearly, use of the spare pairs requires that they be connected all the way from PSE to PD, which may not be the case in some legacy installations. The PoE power limit is 13W per PSE port. A new standard is being discussed which will raise this to about 25W. But don't expect it for a few years and it's primary use is security cameras requiring pan/tilt/zoom. Newer ethernet switches include the PSE function internally, but Midspan Hubs can also be used to insert PoE power in legacy installations. Legacy PDs can also be powered by PoE 'splitters' or 'taps', which pull the power from the ethernet and deliver it to the PD via a short cable. PoE appliances include: Phones Cameras RF ID readers Displays Wireless Access Points Musical instruments The PoE standard is IEEE 802.3af. It was approved about a year ago. There are previous, proprietary PoE schemes from a number of vendors. PoE's -48 V dc is designated as Safety Extra-Low Voltage (SELV). SELV (safety extra low voltage) is a secondary circuit which is designed and protected so that under normal and single-fault conditions, the voltage between any two accessible parts does not exceed a safe value (42.2 V peak or 60 V DC). It is lower than standard telephone network voltage (TNV). -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Monday, November 08, 2004 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [OT] Old Building Needs a New Telephone System Tim Donahue wrote: First, I will admit that I have not worked with PoE before so I'm asking this for my own benefit as well as the OP's benefit. Doesn't PoE require at lest 3 pairs to be availible? I know that pins 1, 2, 3, and 6 get used for ethernet communications and doesn't the power get transmitted over pins 4 and 5? A PoE-enabled connection needs all four pairs. Two pairs for Tx/Rx, one pair for power, one pair for ground. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Asterisk Brazillian Community
Hi Denis, congratulations for the initiative. I would be glad to help. Feel free to contact me PVT. Regards Geraldo -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Denis Galvão Enviada em: sexta-feira, 5 de novembro de 2004 17:55 Para: [EMAIL PROTECTED] Assunto: [Asterisk-Users] Asterisk Brazillian Community Hi all!!! Im proud to announce that we are creating an Asterisk Brazillian Community! We are working hard to bring this wonderful piece of software in our native language, brazillian portuguese. The community will get start after Latinoware 2004 (http://www.latinoware.org) where we will give a lecture about VoIP and Open Source. Im inviting all of portuguese speakers to join with us the Asterisk Brasil community. If you wnat to go to Latinoware, we will give some Asterisk stickers for all of you that want to participate in our community. As soon as possible I will send the wesite URL and other information. For now, if you want to contribute, send me an email. Best regards! Denis Galvão. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users smime.p7s Description: S/MIME cryptographic signature ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: FW: [Asterisk-Users] Need a creative solution - Caller IDanda stupidupstream
Ok. I discovered that this flag will not work, it actually sets the caller
ID to the extension being dialed, ie:
exten = 1235551212,1,Dial(SIP/whatever,15,f)
works perfectly. The caller id will show 1235551212, however:
exten = 1212,1,Dial(SIP/whatever/15,f)
does not work. I believe it tries to set the caller ID to 1212, and
completely ignores what's in the sip.conf file in the callerid= field.
This would work fine for external callers, but if somebody wanted to dial an
internal extension, like 101, it'll try to set the caller ID to 101 and that
won't work. Office users would have to dial the 10 digit number.
This would be fine for home users, but for internal offices this won't work.
I verified that the callerid field was being ignored on forward by setting
what's in the callerid field to 1235551213 and when I placed a normal call
from my voip phone to my cell, the caller ID did show 1235551213; however
when I did CFWDALL on my voip phone to go to my cell phone, I then called
the 1235551212 number with another cell phone, the number that showed up on
my cell phones caller ID was 1235551212;
Help anyone? I hate caller ID.
Ohh, and a side thought, how many of you out there had cell phone usage
triple since you got into VOIP? Hehe.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Rodan
Sent: Monday, November 08, 2004 1:02 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: FW: [Asterisk-Users] Need a creative solution - Caller IDanda
stupidupstream
Hmmm... You're right, I must have missed that option. If this works, I do
apologize for wasting your valuable time. However, do I put this on the
outbound or inbound rule?
This rule:
; Local
exten = _9NXXNXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:1})
exten = _9NXXNXX,2,Congestion
or do I put this on the actual extension of the person who has the CFwdALL
option set, this rule:
exten = 3024,1,Dial(SIP/sales1,20,r)
exten = 3024,2,VoiceMail(u3043)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Monday, November 08, 2004 12:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: FW: [Asterisk-Users] Need a creative solution - Caller ID anda
stupidupstream
Paul Rodan wrote:
Nobody responded so I'm sending this out again. I need help on stopping
the Change caller ID on forward trick that either Cisco or Asterisk
keeps doing. My upstream provider doesn't like it.
This doesn't help?
'f' -- Forces callerid to be set as the extension of the line
making/redirecting the outgoing call. For example, some PSTNs
don't allow callerids from other extensions then the ones
that are assigned to you.
Of course you have to do a show application dial to see all the Dial()
options.
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[Asterisk-Users] MWI Doesn't Turn Off
Anyone having issues with the message indicator lights after CVS-HEAD-07/23/04-13:55:59 ?? For several of my users, our MWI lights do not turn off. Phones are Polycom IP500 and this just started prior to my last update. Should I update to a newer version? I pulled this from the CVS last week so I thought it was newest. Thanks, Wiley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MAX TNT SIP / Asterisk
Have you attempted to use SIP? It's working quite well for me. sip.conf [maxtnt] type=friend host=xxx.xxx.xxx.xxx dtmfmode=inband callerid=MaxTNT maxtnt context=toll-access qualify=yes reinvite=no canreinvite=no disallow=all allow=g729 allow=ulaw extensions.conf (xxx.xxx.xxx.xxx would be the address of your MaxTNT) [toll-trunks] ; ; Outbound 1-nxx-nxx- goes via: PSTN ; exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _1NXXNXX,2,Hangup [local-trunks] ; ; Outbound to nxx- goes via: PSTN ; exten = _NXX,1,Dial(SIP/[EMAIL PROTECTED],60) exten = _NXX,2,Hangup ; [local-access] ; ; Extensions that are this context are allowed to only call local PSTN numbers and other extensions ; include = extensions include = local-trunks ; Access to Local numbers [toll-access] ; ; Extensions that are this context are allowed to call local and long distance PSTN numbers and other extensions ; include = local-access ; Everything local-access has include = toll-trunks ; Access to toll numbers - Darren On Mon, 2004-11-08 at 10:36, James Taylor wrote: Your question indicates that there may be a better way... ??? I want to use the voice mail and extension features of Asterisk, and sometimes there is this NAT problem that Asterisk seems to handle very well. I've been using H.323 with the TNT. Do you have an alternate solution? On Mon, 8 Nov 2004 10:41:31 -0500 (EST), [EMAIL PROTECTED] wrote: On Tue, 2 Nov 2004, James Taylor wrote: I can't get my MAX TNT to register with Asterisk. TAOS 11.0. SIP phone registeration show up in Asterisk like this: sip:[EMAIL PROTECTED] and works. The TNT shows up as: sip:@ip_address. Does anyone have this working? Am I missing something here? Where does the TNT get it's user name? Or, can it work without one? It works without one. Why do you need to register TNT to asterisk anyway? --alex ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MWI Doesn't Turn Off
What does it show in /var/spool/asterisk/voicemail/default/extension/INBOX/ ? Sometimes when my users delete a message or move them around, the sequential order in the INBOX will get thrown off. So the phones light will stay on, because Asterisk can see a file(s) in there, but when they go to access their voicemail, itll say they have no messages, because the voicemail system doesnt see a msg0.wav file, instead there would be a msg6.wav file or something like that in there. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. Siler Sent: Monday, November 08, 2004 2:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] MWI Doesn't Turn Off Anyone having issues with the message indicator lights after CVS-HEAD-07/23/04-13:55:59 ?? For several of my users, our MWI lights do not turn off. Phones are Polycom IP500 and this just started prior to my last update. Should I update to a newer version? I pulled this from the CVS last week so I thought it was newest. Thanks, Wiley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iPeya iPHONE-1001M?
Anybody know anything about this phone: http://www.ipeya.com/SIP_Phone_1001.htm Other phones they sell look like Grandstream phones. Could this be Grandstream's new phone? Thanks, John -- John Gray [EMAIL PROTECTED] AgoraNet, Inc. (302) 224-2475 102 E. Main Street, Suite 303 (302) 224-2552 (fax) Newark, De 19711http://www.agora-net.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail Macro issue.
Hi, Callback is what I based my script on. The problem I am having is when someone leaves a messages and then hangs up, the rest of the macro does not continue to run. If after I leave a message I hit # it works perfect. Any ideas? John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Monday, November 08, 2004 1:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Voicemail Macro issue. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Sent: Monday, November 08, 2004 10:39 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Voicemail Macro issue. Hi, Anyone know how to get voicemail to continue running the next exten in the dialplan when a user hangs up. If a user hits # after leaving a message instead of hanging, up it works. I am trying to do a call back macro and when users hangup after leaving a voicemail the rest of my macro does not run. Any help would be appreciated. Thanks John Bittner Simlab.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This might work for you http://www.voip-info.org/wiki-Asterisk+tips+callback It will not work for us, because we need a repeated call out until the message is picked up, so I have posted a bounty, if the feature I am looking for interests you enough to contribute please add your contribution to the bounty, at some point it will be attractive enough for a coder to do the work. Details on the bounty are here http://www.voip-info.org/tiki-index.php?page=Asterisk%20boun ty %20outcall %20notification%20application ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting jitterbuffer in with iax
On Mon, 8 Nov 2004, Mamadou Lamine KA wrote: Hello everybody; I would like to know the parameters on which depend jitterbuffer in iax.conf. Is there some kind of formula to set the correct values? Thanks in advance for any help Lamine I'd say that the numbers in the iax.conf.sample are a good balance. You'll also find quite a lengthy explanation of the fields in that sample file. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RES: [Asterisk-Users] Astricon Brazil. Why not ?!
Hello Geraldo and Partners , I can offer a conference room on my * BOX on next friday to give a start on this idea. This conference will be made in Portuguese and will start at 11/10/2004 At 8:00PM If someone is interested , please contact me off list for more information. Best Regards, -Jefferson Carvalho Geraldo Fco. do Espírito Santo Jr. wrote: Hi guys, I am so happy to hear that. I'm agree with Jefferson There a lot of people working with * here in Brazil. I would be happy to help you. I have a idea, what do you think about have a chat some day in this week with the Brazilian guys? Geraldo Santo Osasco-SP-Brazil -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Rodrigo P. Telles Enviada em: segunda-feira, 8 de novembro de 2004 09:58 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Astricon Brazil. Why not ?! -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Jefferson, Jefferson Carvalho wrote: | Hello list , | | I'm looking for partners in Brazil to discuss a possible way | to have in Brazil an Official Conference regards Asterisk. Yes, it will be wonderfull ! | It'll includes a hardware/workshop and tech-seminars. | Would be nice if we could include in this conference , Anatel's | presence and a seminar about the lawful aspects of VoIP in Brazil. | I'm 100% sure that in Brazil , we have enough resources to | become a large and active Asterisk community. :) I'm glad to say that you are right and more, there are a lot of people in Brazil working with Asterisk. Contact me in private if you want. Best regards. | | Best Regards, | | -Jefferson Carvalho | Jeff Networks Consulting Ltda. | Teresina-PI-Brazil | ___ | Asterisk-Users mailing list | [EMAIL PROTECTED] | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | - -- Rodrigo P. Telles [EMAIL PROTECTED] Project Manager Devel-IT - http://www.devel-it.com.br TDKOM Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.0.7 (GNU/Linux) iD8DBQFBj19BiLK8unYgEMQRAqG4AJ0azCrspMj2Ca0m/bc6FERBf2lP6QCfYKEZ oguuMN/B5xP8WofrQppKI6Y= =9dRq -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: FW: [Asterisk-Users] Need a creative solution - Caller IDanda stupidupstream
On 11:39 AM 11/8/2004, Paul Rodan wrote:
Help anyone? I hate caller ID.
I would do something like this:
Set accountcode to the callerid number for each sip ua. In other words if
my callerid for a sip UA was John F. Doe 2025551212, then I would set
the accountcode to 2025551212
Then I would create an context/extension that people would dial for setting
a forward number and include it in the contexts available to the SIP UA:
[features]
exten = 1234,1,Answer
exten = 1234,2,Playback(enter-fwd-number-at-tone)
exten = 1234,3,Read(number,,11) ;set length as you see fit, 11 allows +1
US dialing
exten = 1234,4,Wait(1)
exten = 1234,5,Playback(You-entered)
exten = 1234,6,SayDigits(${number})
exten = 1234,7,Background(press-1-if-correct-2-if-incorrect)
exten = 1234,8,Goto(7)
exten = 4321,1,Answer
exten = 4321,2,DBPut(${ACCOUNTCODE}/FEATURE/FORWARD=0)
exten = 4321,3,Playback(forwarding-disabled)
exten = 4321,4,Hangup
exten = 1,1,DBPut(${ACCOUNTCODE}/FEATURE/FORWARD=1)
exten = 1,2,DBPut(${ACCOUNTCODE}/FEATURE/FWDNUMBER=${number})
exten = 1,3,Playback(thankyou)
exten = 1,4,Hangup
exten = 2,1,Goto(1234,2)
Then for inbound calls I which go to the SIP UA, I would check forward
status:
[macro-ring-sip-ua]
; ARG1 is sip extension, ARG2 is timeout, ATG3 is options, ARG4 is callers
callerid
exten = s,1,DBGet(FWDSTATUS=${ARG1}/FEATURE/FORWARD)
exten = s,2,GotoIf($[${FWDSTATUS} = 1]?s,20:s,10)
exten = s,102,NoOp(No DB entry FORWARD for ${ARG1})
exten = s,103,Goto(s,10)
exten = s,10,Dial(SIP/${ARG1},${ARG2},${ARG3})
exten = s,11,Voicemail(u${ARG1})
exten = s,12,Hangup
exten = s,111,Voicemail(b${ARG1})
exten = s,112,Hangup
exten = s,20,DBGet(FWDNUM=${ARG1}/FEATURE/FWDNUMBER)
exten = s,21,SetCallerID(${ARG4} ${ACCOUNTCODE})
exten = s,22,Dial(local/[EMAIL PROTECTED])
exten = s,121,NoOp(No DB entry for FOWARDNUMBER for ${ARG1})
exten = s,122,Goto(s,10)
The idea here is that you are sending out the original caller's ID as the
TEXT field and your callerid as the number field.
Please forgive any typos above, I did this in a few minutes. It should at
least point you in a good direction if this solution is of interest to you.
-Chris
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[Asterisk-Users] Same Extensions in Multiple contexts
Hi For a customer, I am trying to setup 3 different companies on one asterisk box, and I need to assign extension 200 in three different companies. I was using different contexts, but was unable to get it to work. So, my basic question is - In Asterisk, Can we have same extension number in different contexts? For example: [Context_company_1] exten = 200,1,,, [context_company_2] Exten =200,1,.. [context_company_3] Exten =200,1,.. Thanks Uma Pandey ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [OT] Old Building Needs a New Telephone System
It actually uses 2 wires for positive and 2 wires for ground/negative? So it's combing 2 wires (instead of 1) to deliver more power? Which 2 are positive and which 2 are negative/ground? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin P. Fleming Sent: Monday, November 08, 2004 2:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [OT] Old Building Needs a New Telephone System Tim Donahue wrote: First, I will admit that I have not worked with PoE before so I'm asking this for my own benifit as well as the OP's benifit. Doesn't PoE require at lest 3 pairs to be availible? I know that pins 1, 2, 3, and 6 get used for ethernet communications and doesn't the power get transmitted over pins 4 and 5? A PoE-enabled connection needs all four pairs. Two pairs for Tx/Rx, one pair for power, one pair for ground. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Old Building Needs a New Telephone System
Cat3 - which used to be called D Inside Wire (DIW) *is* the wire spec'd in the 10baseT IEEE standard. The existing wire plant is currently to the 10baseT standard., at least as far as the wire goes. (It was originally invisioned that 10bt and analog/digital voice would be running in the same 4 pair cable) Also, the 10bt standard states that 100 meter runs are typical using DIW. There really is no distance standard with 10baseT, only that it typically will run 100m using DIW. PGE, back in the late 80's had a working 10bt run at The Geysers in California of over 500 feet using DIW (ATT Starlan hubs w/ receive threashold set below the standard, BER was still within spec). That being the case, will DIW support 100baseT? The answer is sometimes it will, sometimes it won't. I've seen 200 foot runs of DIW running 100baseT and BER is within spec. The bottom line is you might think of *testing* if baseband ethernet (10, 1000, whatever) will run using the existing wireplant before attempting some dsl/dsl like technology. It would be the least expensive route BTW: Tut make a great product. You might also look at Patton's Ethernet Extenders, another dsl like product that's cheap -JB Hawaii Joe Greco wrote: So how can I do this? Can I use RS485 adapters to get ethernet to each office via the two pair? What kind of data rate can I get with RS485, and would it be half- or full-duplex? Would wireless work in a steel building? Is there some other technology that can be used? What's all this about RS485? 10/100 Ethernet is two pair (unless you get something stupid like 100VG). You probably can't get the 100 on any reasonable run of Cat3, but by all means, run 10. We've done it in the past over fairly long distances, thanks to full duplex you need not worry about the collision domain issues. Wireless might be an option but it's also a security nightmare. ... JG ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [OT] Old Building Needs a New Telephone System
Paul Rodan wrote: It actually uses 2 wires for positive and 2 wires for ground/negative? So it's combing 2 wires (instead of 1) to deliver more power? I believe so, although apparently there is a configuration where the power is present on the data wires instead... I've never seen that though. Which 2 are positive and which 2 are negative/ground? I do not know for sure... I'm looking at a page that says 4/5 are positive, and 7/8 are negative. It must be correct, because it's a page on how to build your own injectors/splitters :-) http://www.nycwireless.net/poe/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cordless vs Wireless phones
The WiSIP phone supports WEP 128 encryption. Not sure if it supports WPA encryption, but that'd be your best bet. I'd use maximum encryption, and separate your AP from your regular network. Just plug an AP into another Ethernet card on your Asterisk server. The phones only need to talk to the Asterisk server, no internet access or anything else. So even if somebody spent the time it'd take to break the encryption, they don't get internet or access to workstation or servers or anything. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kubat, Philip Sent: Monday, November 08, 2004 2:19 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Cordless vs Wireless phones We currently have an Asterisk installation and need to add cordless / wireless phones. Requirements are these phone need to be equals to the wired devices, i.e. dedicated buttons for hold, transfer, etc. , e.g. not an ATA connected analog phone cordless phone. Was thinking of using 802.11b SIP phones (etc), but this opens up all the security concerns of 802.11 and the network. Do any of these phone support VPNs? Have to isolate the WLAN from the LAN. If not is there a SIP (or any other Asterisk channel) device that is a cordless phone. Some things like combining an ATA w/a cordless phone? But as one device with all the digital features? Thanks! Phil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to get Stable 1.X via CVS
What would one enter to get the stable or 1.x version of Asterisk and associated modules via CVS? I've googled and wikkied but I'm using the wrong terms or asking the wrong questions. TIA -Nate ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Xten Video Softphone Gets IM, Presence
http://www.eweek.com/article2/0,1759,1708170,00.asp sounds all good, any body get this running with Asterisk yet? Cheers, Dean ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Same Extensions in Multiple contexts
On Mon, 2004-11-08 at 15:43, Uma S. Pandey wrote: Hi For a customer, I am trying to setup 3 different companies on one asterisk box, and I need to assign extension 200 in three different companies. I was using different contexts, but was unable to get it to work. So, my basic question is - In Asterisk, Can we have same extension number in different contexts? For example: [Context_company_1] exten = 200,1,,, [context_company_2] Exten =200,1,.. [context_company_3] Exten =200,1,.. Thanks Uma Pandey This is certainly possible and in fact quite common. The actual extension that gets used just depends on which context you drop the incoming / outgoing call into. Maybe if you give us some more specifics / config examples we can help you out more. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Same Extensions in Multiple contexts
On Mon, 8 Nov 2004 15:43:10 -0500, Uma S. Pandey [EMAIL PROTECTED] wrote: In Asterisk, Can we have same extension number in different contexts? For example: [Context_company_1] exten = 200,1,,, [context_company_2] Exten =200,1,.. [context_company_3] Exten =200,1,.. Sure you can. You'll need to limit callers abilities based on which company they work for. You could do this by including the contexts of the companies. For instance, lets say you have a phone in sip.conf defined. The context=basic-comp1 [basic-comp1] include = voicemail include = context_company_1 This will only allow the caller to match to extension 200 in company 1 and access to voicemail. More information about extensions.conf can be found at http://www.voip-info.org (the wiki), http://www.asteriskdocs.org (chapter 5) and http://www.google.com. HTH, Leif Madsen. http://www.asteriskdocs.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MWI Doesn't Turn Off
Interestinging From my voicemail.conf, my context where I define my mailboxesin this file is [sip] In the sip.conf I have [EMAIL PROTECTED] Changed that to [EMAIL PROTECTED] and it seems to work better now. Thanks! Wiley From: Paul Rodan [mailto:[EMAIL PROTECTED] Sent: Monday, November 08, 2004 12:58 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] MWI Doesn't Turn Off What does it show in /var/spool/asterisk/voicemail/default/extension/INBOX/ ? Sometimes when my users delete a message or move them around, the sequential order in the INBOX will get thrown off. So the phones light will stay on, because Asterisk can see a file(s) in there, but when they go to access their voicemail, itll say they have no messages, because the voicemail system doesnt see a msg0.wav file, instead there would be a msg6.wav file or something like that in there. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wiley E. SilerSent: Monday, November 08, 2004 2:20 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] MWI Doesn't Turn Off Anyone having issues with the message indicator lights after CVS-HEAD-07/23/04-13:55:59 ?? For several of my users, our MWI lights do not turn off. Phones are Polycom IP500 and this just started prior to my last update. Should I update to a newer version? I pulled this from the CVS last week so I thought it was newest. Thanks, Wiley ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
