RE: [Asterisk-Users] SIP extension calls itself intermittently
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lists Pleasants Sent: 05 November 2005 01:59 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP extension calls itself intermittently Intermittently Ill get calls from my only SIP extension to itself via the Zap/1. I have no clue and have found nothing online. I have listed my configurations and a sample of the console messages I see why debugging. Right now it only happens to the 6000 extension. Any assistance is appreciated. Thanks, Chip -- Starting simple switch on 'Zap/1-1' Nov 4 14:00:54 WARNING[4156]: chan_zap.c:5476 ss_thread: CallerID returned with error on channel 'Zap/1-1' -- Executing Wait(Zap/1-1, 2) in new stack -- Executing Answer(Zap/1-1, ) in new stack -- Executing Dial(Zap/1-1, SIP/6000|20) in new stack -- Called 6000 -- SIP/6000-3d34 is ringing == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/1-1' -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (from-pstn, h, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' bart*CLI = Chip, The output above looks to me as an incoming call. I think you would have seen the SIP extension calling on ZAP/1-1 if initiated from your end, and would not call in as the line was busy. I get this now and again in the UK, usually in an evening time when the Telco do an auto check of line status. Regards Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Access to trunks
Bails wrote: - Are there any configuration options to allow certain sip/iax accounts to dial out over specific trunks, and also to stop them dialing out over other trunks. Thanks in advance Bails = Bails, Set the extensions to use certain context's. Example: - 1234, 1235, 1236 use context1 which dials out on ZAP/1 1237, 1238, 1239 use context2 which dials out on ZAP/2 etc. Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
Wolfgang wrote: - I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) == You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help, please help -- IAX2 softphone to server on LAN
David: Also port 1:2 is a good idea to forward to the server as well.. Only needed for SIP. 4569 is all that is required for IAX2. David, Yes, I've also forwarded port 4569 to the server. Since the router is forwarding to the server, I cannot forward it to the client as well -- however, as the client isn't going out past the LAN, it shouldn't matter... unless there's something else going on that I don't know about. Thanks Wolfgang You might try: - [2278] type=friend secret=** host=dynamic context=clientcon --- David J Carter [EMAIL PROTECTED] wrote: Wolfgang wrote: - I've already sunk several hours into this without any real progress, so I'd really appreciate any help My task is simple -- establish a connection between a softphone on XP ProSP2 to a Asterisk server on Linux FC4 over a LAN through a Netgear router. The server will then go out to a PSTN termination service. Thus far, the PSTN termination connection works fine -- I've opened up 4569 with iptables, and forwarded 4569 to the server IP. I am not, however, having any luck connecting the softphone to the server. I can telnet, ftp, and http to the server, but not IAX2. Iaxping times out, registration by Idefisk and Firefly also times out. The server fails to see the client as well. Here's a portion of my iax.conf: [client] type=friend username=client secret=** host=192.168.1.40 context=clientcon and extensions.conf: [clientcon] exten = 2278,1,Dial(IAX2/client) == You say you have 4569 configured in iptables, what about the netgear router? Have you port forwarded 4569 there? Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk/Xlite/ISA 2000 Config
Leigh Wrote: - I'm running an Asterisk 1.09 box, and a W2k Server with ISA 2000 I would like to configure ISA to allow SIP calls. I've configured the router to pass port 5060 (TCPUDP) to the ISA server, and ports 1-10020 (TCPUDP) to the ISA server. My question is, how do I configure ISA? I presume I need to define protocol definitions and a server publishing rule, however I am confused on how to do this. When adding a UDP protocol definition there are many direction options: Receive Send Receive/Send Send/Receive Any help, greatly appreciated! Thanks Leigh = Leigh, I hope you have better luck than me. I can't seem to open just one UDP port for IAX2. I just come to an abrupt stop every time. Regards Dave ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP make outside call
David,
Shouldn't the [outgoing] be exten = 9.,1,Dial(ZAP/3 ... etc
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David H
Sent: 30 September 2005 17:52
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP make outside call
Hi,
I am can make local extension to and from SIP X-Lite
softphone, but I can't dial out using X-Lite but local
analog works just fine. Here are my conf files any
idea?
Thanks,
David
my sip.conf
[general]
bindport=5060 ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0; IP address to bind
to (0.0.0.0 binds to all)
allow=all
[3000]
type=friend
allow=all
username=3000
secret=my_passwd
host=dynamic
context=sip
dtmfmode=rfc2833
my extension.conf
[globals]
davidHand=Zap/1
davidVoicemail=[EMAIL PROTECTED]
johnHand=Zap/2
johnVoicemail=[EMAIL PROTECTED]
davidout=Zap/3
johnout=Zap/4
[internal]
exten = 1000,1,Dial(${davidHand},10,r)
exten = 1000,n,Voicemail(u${davidVoicemail})
exten =
1000,n,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten = 1000,n,Wait(1)
exten = 1000,n,Hangup()
exten = 1000,102,Voicemail(b${davidVoicemail})
exten = 1000,103,Hangup()
exten = 2000,1,Dial(${johnHand},10,r)
exten = 2000,n,Voicemail(u${johnVoicemail})
exten =
2000,n,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten = 2000,n,Wait(1)
exten = 2000,n,Hangup()
exten = 2000,102,Voicemail(b${johnVoicemail})
exten = 2000,103,Hangup()
exten = 3000,1,Dial(SIP/3000,20,tr)
exten = 3000,n, Bye()
exten =
i,1,Playback(/var/lib/asterisk/sounds/invalid)
exten = i,2,Goto(incoming,s,2)
exten =
t,1,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten = t,2,Hangup()
[outgoing]
ignorepat = 9
exten = 9,1,Dial(Zap/3)
exten = 9,n,Congestion()
exten = 9,n,Hangup()
[voicemail]
exten = 2828,1,VoiceMailMain()
exten = 2828,n,Hangup()
[incoming]
exten = s,1,Answer()
exten =
s,2,Background(/var/lib/asterisk/sounds/vm-enter-num-to-call)
include = internal
[sip]
include = internal
[default]
include = internal
include = outgoing
include = sip
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RE: [Asterisk-Users] Cannot figure out why calls from myAsterisk appear to be fr
Do you use BT for you outgoing calls? Or are you using another provider? I have one customer who uses another provider and there calls come to me with some strange CLI numbers. It seems to be they break out where the best rates are at that time. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ian Bonham Sent: 29 September 2005 15:59 To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Cannot figure out why calls from myAsteriskappear to be fr Not sure about the Digium, but I can tell you +34 is Spain, if that helps you track anything down? I assume you've tested the line with a normal phone to make sure it's not a telco fault? Ian From: Angus Comber [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cannot figure out why calls from my Asteriskappear to be from country code +34? Date: Thu, 29 Sep 2005 15:32:39 +0100 Hello When I dial out from my Asterisk (using Digium analog TDM04B card over pstn line), calls appear to be from +34rest of number I am in UK which is +44 so cannot work out why seeing +34. In my zapata.conf I have: loadzone = uk defaultzone = uk I can't find any country specific stuff in any other conf files. Any ideas how I can correctly set so that calls from my asterisk do not have +34? Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] newbie uk questions...
Darrell, You could try talking to Telappliant, (in London like yourselves), I use them for one of my connections and have found them very good. ISDN is the best way to go if you are looking for your own PSTN connections and to cut down on hardware in the machine I would be looking at an ISDN-30 as only one card is required for up to 30 lines. They say the break point for ISDN-2e to ISDN-30 is 8 lines here in the UK. Alternativly look around at some of the UK companies offering VOIP services it may be quicker and cheaper in the long run to get them to sort it all out for you. Two spring to mind www.telappliant.com and www.holdentel.com . Hope this helps. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Darrell Berry Sent: 13 March 2005 11:21 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] newbie uk questions... hi: Just starting out with *, and I'm planning to heed the advice to start simple and small, but the goal i'm aiming for eventually is: *-based pbx for 10-20 seat small business, based in the UK. Users will have PoE SIP hardphones. So far so good, but two questions, both UK-specific, relating to connection to the outside world (PSTN or VoIP): - are there any UK-based VoIP providers targetting small business users: by which I mean support for multiple simultaneous connections in and out on the same DDI (to simulate traditional multi-channel ISDN PBX capabilities), and guaranteed SLAs/professional support? If so, has anyone dealt with any of them and do you have any recommendations (either for or against?). This includes ISPs getting into the VoIP arena. - failing that, what my options for *-compatible, UK-legal interconnections between a *-based PBX and UK PSTN? I'm looking for more channels than I will get from ISDN-2e, but less than ISDN-33 (probably): enough for say 4-8 simultaneous incoming/outgoing calls. I admit this is the area I'm least clear on! Even better: has anyone actually implemented either of these scenarios in the UK? Any feeeback/cheatsheets? Thanks - Darrell ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file
I have used the Draytek 2600V router in a few locations where only 1 or 2 phones are required. The router has 2 FXS ports and can be used locally to an * box or via the VPN to a remote * box. The VPN built into the routers just works, and I have 1 user who has had 3 VPN circuits up and running now for 6 months solid. Not bad in this day and age for an ADSL to stay functional for so long without interruptions. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anton Krall Sent: 05 March 2005 04:56 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk behind NAT -- SIP config file The VPN approach might resolv a lot of nat issues I guess... Depending on the scenario I guess.. You could put another * box inside the second nat and interconnect using IAX, or if using a single phone, just use your setup, and finally, if using 2 or more phones and cant put a second * box, well, the vpn solution, I wonder how to do it if you have ATAs and nost softphone on the second NATted LAN.. Well... In time I guess :) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rudolf Ladyzhenskii Sent: Viernes, 04 de Marzo de 2005 10:20 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk behind NAT -- SIP config file Yes, only port 5060. If you do not forward 5060, you can not call this phone from outside. Seem to work OK without other ports being forwarded. You mean on the remote sip phone firewall? What if there arem ore than 1 sip phone on that network behidn that firewall? Then you are in trouble. Asterisk only sees single public IP address. As far as it concerns there is only single phone out there. If you get multiple phones working, let me know. Another option, I think, may be using VPN, but I have not tried that. Then you can potentially have remote SIP phones to be on the virtual network. Don't you need to forward ports 1-2 for voice? Or does the sip phones just open up the ports from inside (by doing the in to out calls and keep alives)? I have mot tried to sniff on the traffic in details. I think, other ports are opened in responce to connection on port 5060. The only port listens at is port 5060. Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Digium hardware in the UK ?
Nigel, Should really be on the biz list for this, but Telappliant sells Digium hardware. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nigel Taylor Sent: 05 March 2005 21:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Digium hardware in the UK ? Can anyone recommend a source of Digium hardware in the UK ? Thanks in advance Nigel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Has anyone got early dial working on asterisk ?
Nigel, I have bugetone phones working with 2, 3, 4 + extension numbers. Check you config's, or post them here and lets see if we can find the problem. Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Nigel BurgessSent: 04 March 2005 20:55To: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Has anyone got early dial working on asterisk ? Has anybody got the Early Dial feature working on Asterisk with a grandstream phone ? I can do two digit dials eg 12 and it works fine. When I press a 3rd digit I get a busy response. I did add the auth=plain text in my sip.conf file but to no avail. I have also done the redirect line to make UDP port 0 redirect to port 5060. Anyone had any success ? Cheers Nigel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Getting Polycom IP500 to talk to Asterisk - um... Newbie question :)
*CLI sip show peers Name/usernameHostDyn Nat ACL Mask Port Status 176polycom 192.168.0.176 255.255.255.255 5060 Unmonitored 175polycom 192.168.0.175 255.255.255.255 5060 Unmonitored Added to sip.conf: [175polycom] type=friend host=192.168.0.175 defaultip=192.168.0.175 dtmfmode=inband mailbox=175 context=sip callerid=I am Don progressinband=no ;polycom's seem to have trouble with the default progressinband=never [176polycom] type=friend host=192.168.0.176 defaultip=192.168.0.176 dtmfmode=inband mailbox=176 context=sip callerid=I am a jerk progressinband=no ;polycom's seem to have trouble with the default progressinband=never Don, I would get rid of the number/name combo and use just a number. [175] type=friend host=192.168.0.175 defaultip=192.168.0.175 dtmfmode=inband mailbox=175 context=sip callerid=I am Don progressinband=no ;polycom's seem to have trouble with the default progressinband=never In extensions.conf in your [sip] context add exten = _17X,1,Macro(stdexten) exten = _17X,2,Hangup Regards Dave ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Outbound call on TDM400P
Guy,
I
think what Lyle meant was to put a wait as in dial -- wait ---
number.
Therefore the line is seized and then after a wait the number is
dialled.
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of Guy C.
GuckenbergerSent: 27 February 2005 22:17To: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: RE:
[Asterisk-Users] Outbound call on TDM400P
ok so I put the wait in and still have the same
results.
Extensions.conf
exten = s,5,SetCallerID(${OUTCID})
exten = s,6,Wait(2) -I
added this
exten = s,7,Dial(${OUT}/${ARG1})
exten = s,8,Congestion
exten = s,107,Macro(outisbusy)
Im still only getting out every
few calls. Any other suggestions?
Thanks
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lyle
GieseSent: Sunday, February 27, 2005 1:11 PMTo: Asterisk
Users Mailing List - Non-Commercial DiscussionSubject: Re:
[Asterisk-Users] Outbound call on TDM400P
Put a 'w'ait in your dial command. * is
probably dialing too quickly after going off-hook.
Lyle
- Original Message -
From:
Guy C. Guckenberger
To: asterisk-users@lists.digium.com
Sent: Sunday, February 27, 2005 12:00
PM
Subject: [Asterisk-Users] Outbound call
on TDM400P
Ok all here is a
strange one..
I have a TDM400P
with 3 fxo modules. I can very rarely make an outbound call to the
PSTNabout once every 10 tries. However if I use a analog phone
pluged into the same phone line as one of the tdm channels say channel 4,
and I place the analog phone off hook and then place a call via asterisk ,
it work everytime. It seems like the TDM400p is having trouble
grabbing the outbound circuit. I have tried this on all three
fxo modules and get the same results. Inbound calls work fine as do
SIP calls. Anyone else run into this? Maybe I have bad
hardware?
Thanks
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[Asterisk-Users] Strange problem with h323
All, I have downloaded and installed openh323 as per the documentation. When the machine now reboots safe_asterisk just keeps restarting. If I start another session and just load asterisk -vvvgc asterisk loads. If I enter noload chan_h323.so in the modules.conf then safe_asterisk will kick in. Not 100% on Linux yet but I have added the environment variables info into /etc/profile so they would load each time a reboot takes place, (thought this is the right place). If I do export, the list doesn't show the environment variables, so I assume I have added them in the wrong place. This I assume is why h323 is failing. Anyone point me in the right direction as to where to load these variables, so they load every time? Thanks Dave ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel Red Alarm
It means for some reason you lost your CO line for 10 Seconds. Either someone pulled the plug out by mistake or the Exchange line went away for 10 seconds. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anton Krall Sent: 23 February 2005 09:35 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Zaptel Red Alarm Guys.. I just saw this for the first time... I did some google and wiki without any luck.. what does a red or yellow alarm mean in zaptel? Feb 23 02:54:16 WARNING[16890]: chan_zap.c:5865 handle_init_event: Detected alarm on channel 2: Red Alarm Feb 23 02:54:24 NOTICE[16890]: chan_zap.c:5860 handle_init_event: Alarm cleared on channel 2 This just happened by itself.. __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Menu Selections Only Work Internally
In your [mainmenu] use the include = context_for_internal_numbers, or at
least the ones you want peaple to call.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Philip
Siegrist
Sent: 11 February 2005 15:58
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Menu Selections Only Work Internally
yes. it get's to the Menu prompt which is defined under [MainMenu].
The input buttons simply do not work.
On Fri, 11 Feb 2005 09:06:26 -0600, Jay Milk [EMAIL PROTECTED] wrote:
Does your incoming context include the MainMenu?
-Original Message-
From: Philip Siegrist [mailto:[EMAIL PROTECTED]
Sent: Friday, February 11, 2005 8:17 AM
To: Asterisk-Users@lists.digium.com
Subject: [Asterisk-Users] Menu Selections Only Work Internally
All,
Funny problem. During my greating, the menu selections only
work if one calls from an internal sip line. The greating
plays for all including calls over the t1. But pressing 9 for
directory or any other mapped button will only work if I call
from inside. If I arrive to the menu from an outside line SIP
or POTS pressing the button does nothing. Any ideas?
extensions.conf
--
[MainMenu]
exten=s,1,Answer
exten=s,2,Wait(1)
exten=s,3,Background(main-menu)
exten=_3XX,1,Goto(sip,${EXTEN},1)
exten=0,1,Goto(sip,301,1)
[sip]
;Main Number
exten = 300,1,Goto(MainMenu,s,1)
--
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RE: [Asterisk-Users] Asterisk connected to pbx
How do you want Switch to appear to Asterisk. 1. As an extension. Then use an FXS connection to a CO line input. 2. As a CO line. Then use an FXO connection to an Extension output. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 09 February 2005 05:25 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk connected to pbx I want to connect an asterisk box to a typical pbx switch. What kind of interface i must use: FXS or FXO?And why? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk with Multitech MVP400
Luis, Am I right in thinking that the MVP400 is the non SIP MultiTech box. The SIP version I think is the MVP410. You could load the H323 stack on the box and use H323 to connect to Asterisk. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 07 February 2005 20:21 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk with Multitech MVP400 Hi, I'm new in this telephony stuff, I have Asterisk (default config) installed in one machine (192.168.0.3), and I have a Multitech VoIP gateway running with a phone in port 1 FXS. (192.168.0.14) I've read a lot of info in many places, but I can't figure what I have to do to make that phone ring, actually I don't have a clear Idea how it must be. The goal (my boss' one) is that I have to make that box be a gateway between the FXO interface from outside and Asterisk. But first at all, It is true that I have to edit the sip.conf file? What I have to do? As I said I'm new in this thing. Thanks to all. Luis. _ Do You Yahoo!? Información de Estados Unidos y América Latina, en Yahoo! Noticias. Visítanos en http://noticias.espanol.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] inter asterisk
One thing I do on remote sites is set up a soft phone so I can call myself, this proves out the link and quality before anything else. DIAX id good for this as you can connect to multiple sites, also good to see if you have problems before anyone else calls you to say there is a problem. It also helps in cases like this, if your return quality is good then the possible fault lies with the ZAP interface. Process of elimination, works for me every time. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ousmane Doukara Sent: 06 February 2005 08:46 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] inter asterisk Hi, I am trying to forward calls to another * server with IAX Here is What I want to Do 1- Call SERVER1, let say at 51412345678 2- SERVER1 should transfer the call to SERVER2 in a remote location 3- SERVER2 Receive the call and transfer it to the PSTN number. I have one X100P card on each machine. What is happening is that when the remote party picks up the phone, all he can hear is a weird sound. CONFIGS: SERVER1: zaptel.conf - ~ [channels] ~ language=fr ~ context=montréal ~ signalling=fxs_ks ~ usercallerid=yes ~ callwaiting=yes ~ threewaycalling=yes ~ transfer=yes ~ cancellforward=yes ~ echocancel=yes ~ echocancelwhenbridged=yes ~ echotraining=yes ~ relaxdtmf=yes ~ busydetect=yes ~ busycount=4 ~ callprogress=yes ~ group=1 ~ channel=1 -- (same for SERVER2) IAX.conf ~ [general] ~ bindport=4569 ~ delayreject=yes ~ language=fr ~ allow=all ~ jutterbuffer=no ~ register = username:[EMAIL PROTECTED] ~ tos=lowdelay ~ autokill=yes ~ ~ [quebec] ~ type=friends ~ username = username ~ password=password ~ context=montréal ~ host=Dynamic ~ secret = password ~ disallow = all ~ allow=ulaw ~ allow=gsm extensions.conf --(Same for SERVER2 but no registration) ~ [general] ~ static=yes ~ writeprotect=yes ~ autofallthrough=yes ~ [montréal] ~ exten=s,1,Answer ~ exten=s,2,Playback(message-transfer) ~ exten=s,3,Dial(IAX2/username:[EMAIL PROTECTED]/[EMAIL PROTECTED] al) ; always the same number ~ exten=s,4,Hangup My remote server receive the call, answer the line and then Dial(ZAP/1/51412345678). So far so good. But when 51412345678 pickup the phone, all she can hear is a weird sound. What am I doing wrong ? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten = s,1,Answer
exten = s,2,VoiceMail2(${EXTEN:1})
exten = 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a u or b based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]four more digits
If I hard code the following rules then calls get forwarded
as expected.
exten = u67501,1,VoiceMail2(${EXTEN})
exten = #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten = _[ub][3678].,2,VoiceMail2(${EXTEN})
exten = #,2,Hangup
Thanks
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RE: [Asterisk-Users] Help with extensions
Steve,
Sorry bum information. Line 2 should read: -
exten = s,2,VoiceMail2(${EXTEN})
Don't need to strip the first digit as this is either u or b already,
(Unobtainable or Busy).
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of David J
Carter
Sent: 06 February 2005 12:30
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Help with extensions
Steve,
I haven't tried this but can't you do something like.
[from-proxy]
exten = s,1,Answer
exten = s,2,VoiceMail2(${EXTEN:1})
exten = 3,3,Hangup
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve Blair
Sent: 06 February 2005 12:14
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Help with extensions
Hello:
I'd like some help with defining extension rules. I want calls arriving
at Asterisk from my SIP proxy to be sent directly to voicemail. I'd
also like the appropriate greeting played when the call gets to
voicemail.
My proxy prefixes the extension with a u or b based on
SIP response codes before relaying to Asterisk. So when the
call arrives it is in the format [u|b][3|6|7|8]four more digits
If I hard code the following rules then calls get forwarded
as expected.
exten = u67501,1,VoiceMail2(${EXTEN})
exten = #,2,Hangup
However to save on typing I'd like a general rule. I've tried
the following but Asterisk cannot find the extension with this
set of rules. Can someone explain how what I want can be
accomplished?
exten = _[ub][3678].,2,VoiceMail2(${EXTEN})
exten = #,2,Hangup
Thanks
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[Asterisk-Users] ISDN X-Over
Hi all, I have just been reading an article on the asterisk-doc site about ISDN X-Over cables. The article mentioned the converting of an NT1 to make this possible, has anybody got the information required to modify a BT NT1? Or any information on the BT NT1. Thanks in advance. Regards Dave ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN X-Over
Stefan, Peter, Thanks for the replies guys. I have looked at the web page and will work on it over the weekend. My next step will be to find out hoe the CO lines connect, but that's another project. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stefan Gofferje Sent: 05 February 2005 12:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ISDN X-Over Peter Svensson schrieb: On Sat, 5 Feb 2005, Stefan Gofferje wrote: As far as I know, you just need a second ISDN card and a X-cable. No mods to the NT1 are needed. To build such a cable, just swap the outer pair with the inner pair. Termination and power are needed. Most isdn cards provide neither, but some do. Yes, as I wrote, I was reading to fast... However, termination may be done by simply connecting a terminated multiplug to the card and power is only needed for phones without own power source. I have e.g. a Siemens Gigaset cordless phone connected to my internal port, so I need no power. But anyway, I believed, David was writing about ISDN monitoring, so my answer was inadequate, I confess. May we PLEASE leave out the tar and feathers? Only this time? :-) Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | SuSE Certified Linux Trainer V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX dns lookups
Hi,
Try something like these, works for me.
extensions.conf
[general]
;
static=yes
;
writeprotect=no
;
[globals]
;
CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
;
#include globals.conf ;This includes your conf file with your fqdn's
listed.
exten = _20XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _20XX,2,Hangup
;
exten = _21XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _21XX,2,Hangup
;
exten = _22XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _22XX,2,Hangup
;
exten = _23XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _23XX,2,Hangup
;
exten = _24XX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _24XX,2,Hangup
globals.conf
RMT1=www.domain1.zzz;remote1
RMT2=www.domain2.zzz;remote2
RMT3=www.domain3.zzz;remote3
RMT4=www.domain4.zzz;remote4
RMT5=www.domain5.zzz;remote5
I never reboot even when the DynDns changes.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Liaan vd
Merwe
Sent: 03 February 2005 07:38
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] IAX dns lookups
Hi all
Do any of you know i can force asterisk to lookup ip
addresses for peers and
trunks everytime it tries to make a call.
One of the peers has a dynamic ip and is using DynDNS
to register host. Now
i need to reload asterisk everytime i want to call it
thanks
liaan
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RE: [Asterisk-Users] TDM-400P + CallerID
The only Caller ID phone I can get to work on the TDM card is one with belcore caller ID, the UK callerid phones do not work here. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Lyle Giese Sent: 27 January 2005 17:47 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM-400P + CallerID You really don't need the wait(exten = s,1,Wait(2)). cidsignalling=dtmf cidstart=polarity I am not familar with these options as they don't apply to the US. Here callerid info is sent between the first and second ring as FSK(old modem signaling). It may be that the FXS channels are doing that and your equipment is not setup to do that. Lyle - Original Message - From: Pieter Arentz To: asterisk-users@lists.digium.com Sent: Thursday, January 27, 2005 10:07 AM Subject: [Asterisk-Users] TDM-400P + CallerID Hi, Im just starting out with Asterisk, in combination with a TDM400, filled with 2 FXS on channels 1 and 2, and 1 FXO on 4. Having just started, all I want right now is to be able to answer incoming calls on a phone connected to channel 1. The trouble is the caller id. I have caller id enabled on my line, my phone supports it, and when I connect the phone directly to the line, it works. However, it doesnt work with *. When I call myself (with a cellphone), and I type show channel zap/4-1 in the * console, it shows my cellphone# in the caller id field. Asterisk gets the correct callerid from my line, appearantly. When I type show channel zap/1-1, the caller id just shows s. I have a feeling that this s is the originating extension, seen from the FXS point of view. My phone just shows external call, instead of a number. How do I make * forward the callerid from the incoming call to my phone? --Pieter My zapata.conf: context=buitenlijn signalling=fxs_ks immediate=yes usecallerid=yes cidsignalling=dtmf cidstart=polarity hidecallerid=no callerid=asreceived callwaiting=no callwaitingcallerid=no adsi=no channel = 4 signalling=fxo_ks language=nl usedistinctiveringdetection=no busydetect=yes echocancel=yes echotraining=no channel = 1 channel = 2 My extensions.conf: [buitenlijn] exten = s,1,Wait(2) exten = s,2,Dial(Zap/1,30,t) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Route incoming call on 4 X100P to different Ext.{Scanned}
David, Try something like this:- zapata.conf context=me signalling=fxs_ks channel = 1 ; context=her signalling=fxs_ks channel = 2 ; context=fax signalling=fxs_ks channel = 3 ; context=meandher signalling=fxs_ks channel = 4 extensions.conf [me] exten = s,1,Dial(SIP/0001,30,t) exten = s,2,Hangup ; [her] exten = s,1,Dial(SIP/0002,30,t) exten = s,2,Hangup ; and so on. Regards Dave -Original Message- Hello All, I have 4 X100P cards. I was hoping to have card (line) go to separate ext. i.e. Card 1 (XXX)555-0001 My Ext Card 2 (XXX)555-0002 Wife's Ext Card 3 (XXX)555-0003 Fax Ext Card 4 (XXX)555-0004 My and Wife Ext. This is what I have now and all incoming line rings this one extension. exten = s,1,Dial(SIP/300,10) So what is s . Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cant get Asterisk server talk with IAX
Hi,
Have you got port 4569 open in your NAT/Firewall?
I take it that your extension ranges on the servers are 5000 and 6000 range.
The configs look OK, same as mine, and mine works fine.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Chicku
Sent: 28 December 2004 04:44
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cant get Asterisk server talk with IAX
Hi everyone,
I am trying to connect 2 asterisk servers via IAX, but it just
fails to do so.. I'm using SIP to connect the IP phones on the
LAN at the 2 different physical locations where each server
resides and I'm able to communicate on my LAN via SIP without
any issues. The problem is that I'm unable to make Asterisk
servers talk with each other via IAX..
Here is my issue.
I've got one asterisk server connected directly to the internet
and the other behind a NAT.
The iax.conf file for the one that is directly connected to the
internet is as follows:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on
machine)
delayreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
jitterbuffer=yes
mailboxdetail=yes
#include iax_additional.conf [I'm using AMP as the GUI
interface]
register = 1000:[EMAIL PROTECTED]
[a.b.c.d is the IP address of the router. ie. the server is
behind the nat]
[2000]
type=user
username=2000
auth=plaintext
permit=a.b.c.d/255.255.255.0
host=dynamic
context=fullaccess
My extension.conf is as follows for the server that is directly
connected to internet.:
[fullaccess]
exten = _5XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _5XXX,2,Hangup
exten = _5XXX,102,Hangup
--
Now the iax.conf file for the one behind NAT is as follows:
[general]
bindport = 4569 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on
machine)
delayreject=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
jitterbuffer=yes
mailboxdetail=yes
#include iax_additional.conf
register = 2000:[EMAIL PROTECTED]
[1000]
type=user
username=1000
auth=plaintext
permit=0.0.0.0/0.0.0.0
host=dynamic
context=fullaccess
My extension.conf is as follows for the server that is behind
NAT:
[fullaccess]
exten = _6XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _6XXX,2,Hangup
exten = _6XXX,102,Hangup
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RE: [Asterisk-Users] PrivacyManager 10 digit limit.
I thought the standard for the UK was 11 Digits in length, (save some old 0845, 0800, 0870 numbers), but most of these are transported to normal 11 digit numbers. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Dent Sent: 08 December 2004 14:13 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] PrivacyManager 10 digit limit. Hi Here in the UK telephone numbers vary in length. When PrivacyManager kicks in it seems to only listen for the first 10 digits. Is it possible to have it take any number of digits followed by # to indicate the end of the number? Thanks Mike ___ 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] Billing (itemized) in the UK
Pete,
I am also in the UK and I have added an include in my extensions.conf for
the file listed bellow.
exten = _15X,1,Dial,${TRUNK}/BYEXTENSION
exten = _147X,1,Dial,${TRUNK}/BYEXTENSION
exten = _NX,1,Dial,${TRUNK}/BYEXTENSION
exten = _01.,1,Dial,${TRUNK}/BYEXTENSION
exten = _07.,1,Dial,${TRUNK}/BYEXTENSION
exten = _08.,1,Dial,${TRUNK}/BYEXTENSION
exten = _09.,1,goto(nogo,1)
You dont need a 9 for a line, you couls also add lines for barred numbers
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Peter Hoppe
Sent: 25 November 2004 13:34
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Billing (itemized) in the UK
Thank you very much for the answer! I think it is a good path to look at. I
have had a look through
our paperwork for the present pbx, and I found one document that seemed to
indicate we have to dial
1666extndialled_number
to give the extn info to the telco. The paper is a bit old (1999) and since
then we have changed our
telco, but I guess that this protocol is still valid. This afternoon I will
hook up a recording
device on the line and see which digits are actually dialled when I dial an
outside line. From the
recording I should be able to reconstruct which digits have actually been
dialled by the pbx.
If the protocol is correct, I could construct a dial command such as
exten = _9.,1,Dial(Zap/g1/1666ID${EXTEN:1})
or so - I would just need a way to construct id - and then any caller from
an inside device would
just prepend a '9' before the real number. I probably would also bar simple
'9' dialling to get an
outside line... lets see.
Keep you posted, and so many thanks for all the help!
P
Hi Peter
You need to first of all ask your Telco what mechanism it uses with your
current switch. The most likely ways are
1) Two stage dialling. 1xxx pause PIN exten dialled number
2) access code1xxx exten dialled number
You need to get the specs for this from Your Communications. It is not
clear from the web site...
Asterisk will cope perfectly with either solution - you will just need
to fiddle a bit with the dial plan. Once we know what you have to send
to the telco there are tons of people here who will advise on the Dial
command you should use to achieve what you want.
Rgds
Tim Robinson
Ps. Any reason why you chose to stick with the analogue solution? Is
this just risk mitigation in the early stages? (this is a valid reason,
btw!)
-Original Message-
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Peter
Hoppe
Sent: 25 November 2004 10:54
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] Billing (itemized) in the UK
Hello!
We are located in the UK, and we are planning to replace our old pbx
with an asterisk based pbx. For
outgoing calls our present pbx is connected to three PSTN lines which
all have the same number.
Internally, the pbx caters for quite a few extensions, and each
extension can make outbound phone calls.
Our telecom provider (your communications) gives us monthly itemized
bills that list all of the
calls per extension, i.e. from the bill we are able to tell which
internal extension made what call
to which destination at which date/time, how long this call was in
minutes and how much that
particular call costs.
We would like to reuse the three PSTN lines with the asterisk system,
and at present there are no
plans to utilize other connectiviy (such as ISDN) - we would like to
stick with the three PSTN lines.
My understanding is that when the asterisk system is running we won't
get any itemized bills any
more since the telecom provider has no way of telling from which
extension a call originated.
Questions:
To give the extension information to the telco...
How can I configure Asterisk to do send extension information?
What signalling do I have to provide for outgoing calls to give
extension information the telco?
Is there a standard for sending extension numbers (i.e. do I have to
send some DTMF digits)?
Is there a software / asterisk extension (that works in the UK) that
allows asterisk to send
extension info?
Do I need to buy some equipment that can provide this info to the telco?
Which?
Where could I find more information on that subject?
Thank you very much for your consideration.
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RE: [Asterisk-Users] X100P noise on ADSL line.
HI, Had the same problem a while ago, X100p or a Modem caused the same problems as your getting. Changed the Microfilter and the problems went away. Tested the removed Microfilter and found the High pass filter was Knackered. Mine also showed the error with a phone connected as well but not as bad. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: 26 October 2004 11:35 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] X100P noise on ADSL line. Hi, This may be one for the broadband guru's out there.. I have a single analog line coming into the house.. This line is for my ADSL and home phone.. My Asterisk box uses an X100P card to connect to the analog line.. I have a microfilter on the line etc.. The rest of my phone system works inbound and outbound calls via a VoIP provider over the ADSL line.. The problem I am having is that the X100P seems to introduce a lot of noise on the line when it its connected to the phone socket on the microfilter and this causes the ADSL quality to drop quite badly.. When the X100P is not connected I have a signal to noise ratio of 29dB downstream and 30dB upstream (this stays the same when I connect an analog phone) when I connect the X100P the SNR drops to 12dB downstream and 30dB upstream.. At 12dB I get a large number of CRC errors and errored seconds on the ADSL connection.. Anyone got any ideas why the X100P would cause this kind of deterioration? Only thing I can think of is possibly something to do with ring detection or that its acting on some of the frequencies that are being used by the ADSL.. Thanks for any thoughts.. ___ 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] meetme question
Did
you uncomment the ztdummy in the zaptel Makefile?
Dave
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of christophe
de coninckSent: 21 October 2004 14:03To:
[EMAIL PROTECTED]Subject: [Asterisk-Users] meetme
questionHi,I've just setup a meetme room like in the
config is set as demo, alltough when i try to call it I see
this: -- Executing MeetMe("SIP/christophe-9123",
"1234") in new stack == Parsing '/etc/asterisk/meetme.conf':
FoundOct 21 15:00:23 WARNING[409617]: chan_zap.c:755 zt_open: Unable to
open '/dev/zap/pseudo': No such file or directoryOct 21 15:00:23
ERROR[409617]: chan_zap.c:6663 chandup: Unable to dup channel: No such file or
directoryOct 21 15:00:23 WARNING[409617]: app_meetme.c:227 build_conf:
Unable to open pseudo channel - trying deviceOct 21 15:00:23
WARNING[409617]: app_meetme.c:230 build_conf: Unable to open pseudo
device -- Playing 'conf-invalid' (language
'en') == Spawn extension (default, 8600, 1) exited non-zero on
'SIP/christophe-9123'extensions.conf:exten =
8600,1,Meetme(1234)exten =
8600,2,HangUpmeetme.conf:[rooms]conf = 1234
-- Christophe De Coninck | Zarek K http://www.zarekk.bemailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED]
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RE: [Asterisk-Users] grandstream handytone 286 problem
Christophe, Just for starters try changing your SIP user ID in the 286 to and 4445 and see if they register then. I have several 286's and they all work fine, but I don't use names, just numbers. Regards Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of christophe de coninckSent: 20 October 2004 15:37To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] grandstream handytone 286 problem update:now I get this after i repowered the grandstream handytone 286:*CLI Oct 20 16:34:33 NOTICE[262160]: chan_sip.c:7532 handle_request: Registration from 'sip:[EMAIL PROTECTED];user=phone' failed for '10.0.0.55 On Wed, 2004-10-20 at 15:30, christophe de coninck wrote: hey,i got asterisk running with softphones but now I received a set of grandstream handytone 286's now if I run the setup and configure everything like supposed to be it doesn't work, i hear a ringing tone , after 30secs it hangs up and that's itin sip.conf i have:[4445]secret=4445type=friendusername=christopheallow=allhost=10.0.0.55nat=yes[]secret=type=friendusername=nicoleallow=allhost=10.0.0.56nat=yesand for configuration of my grandstream handytone 286 i got:sip server: 10.0.0.21sip user id: christopheauthenticate id: 4445authenticate password: 4445and as vocoder i got:G729G729G729G729PCMUPCMAPCMUsip registration: yesunregister at reboot: yesanyone know what i could be doing wrong ? -- Christophe De Coninck | Zarek K ___ 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-- Christophe De Coninck | Zarek K http://www.zarekk.bemailto: [EMAIL PROTECTED] mailto: [EMAIL PROTECTED] banner.gif___ 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] Control access to external dialing
Luke, I have a situation like yours, mine is to enable an IAX2 call between two servers and then break out to a trunk. All I have done is added a six digit code in front of the number (eg Birthdate ,210573 or 052173 if in US), and then stripted the six digits before dialing. You only tell the people you want to be able to dial out the six digit code. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Luke Catranis Sent: 20 October 2004 15:55 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Control access to external dialing Wondering if anyone could give me a tip on controlling access under the following scenario. I have an ATA connected to a legacy pbx as a trunk line. I want to control who can make calls on this trunk. I cannot set restrictions on the users via the pbx, so I would like to be able to assign a passcode for people so they can dial out using this trunk line... ___ 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] DIAX 0.9.9b - now multi codec support
Dan, Can I import all the settings from a previous version into 0.9.9b to save re-inputting all the info? Dave = Hi all, Thanks to the great work of Steve Kann on the iaxclient library, now DIAX is able to support the following codecs: - uLaw (still a little ptoblem with the sound in one direction) - GSM - iLBC - Speex You can download version 0.9.9b from the following address: http://www.geocities.com/tdanro/diax/diax099b.zip The help file and the web page is not yet updated (I work on this now). For the latest available help file use the address: http://www.laser.com/dante/diax/diaxhlp.htm Please play with it and send me your feedback. It is not fully tested, so...please be carefull. Thank you for your help and best regards, Dan P.S. The updated source file for the wiax.dll will be available soon on my site. ___ 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] Sending broadcasts to all phones?
I have a Panasonic switch here and it a paging system on the switch. It will output the page message to all phones and also to an RCA (Phono) socket on the side of the switch to a PA amplifier if required to drive a 100Volt line system around a building. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kevin Walsh Sent: 16 October 2004 22:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Sending broadcasts to all phones? Kristian Kielhofner [EMAIL PROTECTED] wrote: Stan Brinkerhoff wrote: A friend of mine has a real panasonic PBX setup at his house, and is able to pick up the phone, dial an extension, and it broadcasts what he says over every phone in his house without the phones having to be picked up. What is this feature called? Would it be possible to set this up with Asterisk given the appropriate phones? (Cisco?) This can be done with Cisco phones and 6.x or 7.x firmware. It is on the wiki. Well, actually, it's not on the WIKI. The WIKI would help you set up a Cisco phone to auto-answer, but that's not all he needs here. The problem is that if you dial phone1phone2 then the first phone to auto-answer will receive the broadcasted call. The other phones in the list will not hear anything. Well, that'd be what I'd expect to happen with Dial(), anyway. Stan seems to be asking for a system where the caller hears a ring tone until all phones (auto)answer, and is then able to speak to them all at once. It'd be kind of like an enforced conference call, but with one speaker and multiple listeners, and with all audio received from the called phones thrown away rather than distributed. It could be done, but would need a new Dial()-based application to do it, I think. Perhaps there's an existing facility that can be used to to do this. If there is then I can't think of it. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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] musiconhold will not start
Try mpg123-0.59r Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Remco Barende Sent: 12 October 2004 22:10 To: Asterisk Users List Subject: [Asterisk-Users] musiconhold will not start I have * running on gentoo. Everything seems to be working fine but musiconhold will not start. When starting * I get these errors, but guess that's not the problem: Oct 12 16:42:12 WARNING[16384]: chan_skinny.c:2584 reload_config: Unable to get our IP address, Skinny disabled Oct 12 16:42:12 WARNING[16384]: chan_oss.c:434 soundcard_init: Unable to open /dev/dsp: No such file or directory When MOH needs to kick in however I get this message: WARNING[294927]: res_musiconhold.c:366 moh1_exec: Unable to start music on hold (class '30') on channel SIP/101-8168 The box has media-sound/mpg123 Latest version installed: 0.59s-r4 I'm not sure why it cannot start the muzak however, the wiki says that a symlink must be created to the binary but the binary is already in place where the symlink should come. ___ 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] Grandstream phone price
$1.64 to the £1 I think this morning so $35 stands. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wolf N. Paul Sent: 11 October 2004 07:40 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream phone price Except that £55 is more like $75-80 and not $35. Regards, Wolf David J Carter [EMAIL PROTECTED] writes: I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Dave Grandstreams are availabe for $65 quanity one, so its not hard to believe that you could get them for $55 for larger quantities ___ 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] Grandstream phone price
Forget the last post, the brain is totally screwed. Must get more sleep. Thanks all for pointing the errors of my conversion, so used to working the other way. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wolf N. Paul Sent: 11 October 2004 07:40 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Grandstream phone price Except that £55 is more like $75-80 and not $35. Regards, Wolf David J Carter [EMAIL PROTECTED] writes: I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Dave Grandstreams are availabe for $65 quanity one, so its not hard to believe that you could get them for $55 for larger quantities ___ 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] Intel Modem vs Digium Cards
I beleive Telappliant in the UK are doing them for £55, ($35) http://www.voiptalk.org/products/index.php?cPath=27 Dave Grandstreams are availabe for $65 quanity one, so its not hard to believe that you could get them for $55 for larger quantities http://froogle.google.com/froogle?q=grandstreamhl=enlr=tab=wfscoring=p Jim James H. Thompson [EMAIL PROTECTED] i am still looking for the elusive $55 grandstreams. ___ 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] Modem[i4l]/ttyI0 sent into invalid extension 's'
Do you have a [remote] context in tour extensions.conf? because that is where the calls are bein sent. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ian Johnson Sent: 23 September 2004 10:22 To: asterisk Subject: [Asterisk-Users] Modem[i4l]/ttyI0 sent into invalid extension 's' G'day, New to Asterisk alert! I have a Netjet card running on linux 2.4.27 kernel using the HiSax module, and trying to use it for incoming/outgoing calls from *. I've tried playing with modem.conf and extensions.conf every which way I can think of, using samples and whatever I can find off the net, and I get the same message everytime I try to dial in. The complete message is: pbx.c:1868 ast_pbx_run: Channel 'Modem[i4l]/ttyI0 sent into invalid extension 's' in context 'default', but no invalid handler modem.conf: [interfaces] context=remote driver=i4l type=autodetect stripmsd=0 dialtype=tone mode=immediate msn=12345678 (not the real number of course) device = /dev/ttyI0 group=1 extensions.conf: Is the sample extensions.conf, at the moment. Any ideas/solutions would be great! Thanks. Ian. -- Nambour Christian College ... Sow to Harvest. http://www.ncc.qld.edu.au __ This email and any files transmitted with it are confidential and intended solely for the use of the addressee. It may contain privileged information that is exempt from disclosure by law. Please note that unauthorised dissemination, copying or accessing of this email and its contents is prohibited and may be unlawful. If you have received this email in error please inform us immediately by telephone on +61 (0) 7 5442 1866. Opinions expressed in this E-Mail are those of the sender and do not necessarily represent the views of Nambour Christian College. Although this email has been created on a machine protected by Anti-Virus software, we cannot be held responsible for any viruses or other material transmitted with or as part of this email. ___ 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] Modem[i4l]/ttyI0 sent into invalid extension 's'
Ian, Contact me off list and we can try and sort it out. [EMAIL PROTECTED] Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ian Johnson Sent: 23 September 2004 12:13 To: asterisk Subject: RE: [Asterisk-Users] Modem[i4l]/ttyI0 sent into invalid extension 's' G'day Dave, Do you have a [remote] context in tour extensions.conf? because that is where the calls are bein sent. I did try putting a [remote] context in, but the error message was indentical. The error seems to be wanting to put it in the [default] context, no matter what I put in modem.conf I'm wondering what the error means by invalid extension 's', am I supposed to have something else, I've tried putting in the calling MSN, as: [remote] and in [default] exten = 12345678,1,Answer But no joy. Sorry about the legal rubbish attached to these e-mails. __ This email and any files transmitted with it are confidential and intended solely for the use of the addressee. It may contain privileged information that is exempt from disclosure by law. Please note that unauthorised dissemination, copying or accessing of this email and its contents is prohibited and may be unlawful. If you have received this email in error please inform us immediately by telephone on +61 (0) 7 5442 1866. Opinions expressed in this E-Mail are those of the sender and do not necessarily represent the views of Nambour Christian College. Although this email has been created on a machine protected by Anti-Virus software, we cannot be held responsible for any viruses or other material transmitted with or as part of this email. ___ 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] Asterisk is not picking up the phone with ax100p card
Rodolfo, Shouldn't it be siganlling=fxs_ls for the x100p ? Where is your channel = 1 What is in your zaptel.conf ? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rodolfo Grave Sent: 15 September 2004 22:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk is not picking up the phone with ax100p card Hi. I have a x100p card installed on my asterisk box... my zapata.conf file includes the following lines: [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 echocancel=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 Basically, the zapata.conf file generated by make samples. Then in my extensions.conf I have this: [default] include = demo And demo is also the included in the sample extensions.conf Now, when I call into my PSTN line connected to the x100p card (which also has a phone attached just to hear the ringing), asterisk is not answering the call. I expected it to go into the [default] context, which should make a welcome speech. I am able to call asterisk from the console and everything goes ok. This is also the lsmod report concerning wcfxo and zaptel when asterisk is not running: wcfxo 8704 0 (unused) zaptel188416 0 [wcfxo] and this is when asterisk is running: wcfxo 8704 0 (unused) zaptel188416 4 [wcfxo] so I assume zaptel is being used by asterisk as expected. Any hints on why asterisk doens't get the call? Thanks in advance. RODOLFO --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0438-1, 14/09/2004 Tested on: 15/09/2004 23:51:53 avast! - copyright (c) 2000-2004 ALWIL Software. http://www.avast.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] Grandstream BugetTone 100 Caller IDshows extension, not incoming Caller ID
Steven,
On mine in the UK the sip.conf entries are like yours but without the
callerid= entry and my CS phones give me the received callerid fine.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steven P.
Donegan
Sent: 12 September 2004 16:55
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] Grandstream Budgetone 100 Caller IDshows
extension, not incoming Caller ID
Eric Wieling wrote:
On Sun, 2004-09-12 at 09:41, Duane wrote:
Steven P. Donegan wrote:
I've looked through the archives - and see questions similar to mine,
but no answers. What, if anything, can be done to get the incoming
Caller ID to be presented on the Budgetone's Caller ID display? In all
other respects the phone+Asterisk seem to be extremely happy with each
other.
What you need to do is strip the alpha caller name from the caller ID,
the 101's can only handle numbers and it's trying to display a name...
I don't think this is the problem. If it was a general problem hundreds
f people would be complaining about this. Put a
NoOp(CALLERID=${CALLERID}) in the dialplan just before the Dial line to
ring the GS phone. What you should see is something like CALLERID=Bob
Dobbs 666 on the console when the NoOp runs. If you see ANYTHING that
isn't in the format of Caller*ID Name calleridnumber. then you have
something messed up in your Asterisk config. As said, the BT101 only
can display Caller*ID numbers, it should generally just throw out the
Caller*ID name. You don't mention what COUNTRY you are in so I don't
know if it's an issue between what your telco sends and what Asterisk
expects. In the USA this is not an issue, in other countries it *could*
be an issue.
I am in the US, and caller ID otherwise works fine (ie on analog
stations it comes thorough just fine).
sip.conf configlet:
[1000]
type=friend
username=1000
fromuser=1000
callerid=Computer Room 1000
host=dynamic
nat=no
canreinvite=yes
dtmfmode=info
[EMAIL PROTECTED]
disallow=all
allow=ulaw
extensions.conf configlet:
[sip-access]
exten = 1000,1,Macro(stdexten,1000,SIP/1000)
The stdexten Macro is the vanilla one from 'stock' Asterisk.
On the console I see all the appropriate caller ID/connection info, and
the Voicemail application definitely emails me the correct stuff - so it
seems it is something being lost between Asterisk/Grandstream...
Thanks for any help - this is on my home PBX - but once it all works I
will be rolling it out as a test at a friendly beta customer :-)
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RE: [Asterisk-Users] X100P question
Gilbert, The phone port is only a loop thru port for the analogue line. It is not an FXS port. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 01 September 2004 09:32 To: Asterisk-Users; Asterisk-Dev-Admin Subject: [Asterisk-Users] X100P question Hi, I have a question regarding X100P card. I have one X100P card in an * box. I have the telco line connected to the line port of the X100P card, and an analog phone connected to the phone port of the X100P card. My question is: How to make ringing the analog phone connected to the phone port when you receive a VoIP call? Thanks. GIBERT Frédéric Mobile: +33 6 72 08 35 16 Fax : +33 1 30 71 39 33 Mail : [EMAIL PROTECTED] Bureau Paris : Ste VIGINETWORKS (Chez CAP retraite) 137, rue vielle du temple 75003 Paris France File: ATT00015.txt attachment: winmail.dat___ 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] Ireland PSTN Number
Hi, Does anyone know of a provider/terminator of Belfast, Ireland telephone numbers? Thanks in advance Regards Dave ___ 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] Problems with X100P
Don't you need a 'modprobe wcfxs' also? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Adam Lewis Sent: 18 June 2004 14:57 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Problems with X100P All, I'm having trouble getting the X100P working. Lsmod shows : zaptel179808 0 I did a . # modprobe zaptel and here is my zaptel.conf (comments omitted) __SNIP__ fxsks=1 loadzone = us defaultzone=us __SNIP__ Here is zapata.conf __SNIP__ [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes group=1 callgroup=1 pickupgroup=1 immediate=no context=sip signalling=fxs_ks callerid=Phone 1 channel=1 __SNIP__ ztcfg -vv gives the following output.. __SNIP__ Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) __SNIP__ Any ideas, Thanks, Adam ___ 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] BT Caller ID - From Patch ?
I have it working with the X100P no problems, on both BT and Telewest lines. Anybody got it working on the TDM400P yet? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Kannaiyan Natesan Sent: 17 June 2004 19:59 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] BT Caller ID - From Patch ? I have the following settings chris. Also i confirmed with BT that caller is enabled on my line. Let me know if I need to modify anything. Thanks. zaptel.conf: fxsks=1 loadzone=uk defaultzone=uk zapata.conf: [channels] busydetect=1 busycount=7 relaxdtmf=yes callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=uk echotraining=yes echocancel=yes echocancelwhenbridged=yes jitterbuffers=4 rxgain=0.0 txgain=0.0 group=1 pickupgroup=1-4 immediate=no context=default signalling=fxs_ks callerid=asreceived channel=1 Kannaiyan - Original Message - From: Chris Stenton [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 17, 2004 7:38 PM Subject: Re: [Asterisk-Users] BT Caller ID - From Patch ? It works fine for me. make sure you only have usecallerid=uk in the config. If you also have usecallerid=yes set it will default to the US style. Make sure you have the uk settings in zaptel.conf. Can you see the callerid with a std phone on the line? Chris - Original Message - From: Kannaiyan Natesan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, June 17, 2004 7:20 PM Subject: Re: [Asterisk-Users] BT Caller ID - From Patch ? [snip] My Zapata.conf: usecallerid=yes ukcallerid=yes Change those two lines to simply usecallerid=uk. I changed as you said and restarted asterisk. Still doesn't work. -- Starting simple switch on 'Zap/1-1' Jun 17 19:24:48 ERROR[262160]: chan_zap.c:4759 ss_thread: zt_get_history failed: Inappropriate ioctl for device -- Executing MySQLput(Zap/1-1, cid/cid=s) in new stack -- mysqlput: family=cid, key=cid, value=s -- Executing Dial(Zap/1-1, SIP/12345|20|tr) in new stack -- Called 12345 -- SIP/12345-810a is ringing == Spawn extension (default, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' And yes, the patches work well. Still I didn't get it to work. Do I need to take care of any other settings? I applied the patch from the bug report. Kannaiyan -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[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 ___ 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] Background Playback fails
You haven't made my mistake and forgotten about case sensitivity in Linux have you. I had the same problem when I called mine Mainmenu and put mainmenu in the dialplan. Regards Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Tim GuySent: 11 June 2004 12:55To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Background Playback fails It worked! Cool I assumed that as the demo sounds didnt have paths, mine wouldnt need them either. Thanks umar -Original Message-From: usedcanon [mailto:[EMAIL PROTECTED] Sent: 11 June 2004 12:40To: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Background Playback fails have tried specifying the full path ? Umar
RE: [Asterisk-Users] IAX Won't Pass Caller ID
Hi, I had the same problem until I changed iax.conf to not have a callerid= field in it for the context you are using. All I have now is. [guest] type=user context=default I have several servers all talk to each other, and get caller/extension ID from them all. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Blackman Sent: 08 June 2004 04:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAX Won't Pass Caller ID Hi, We have to servers set up in two different networks. We are able to connect calls via IAX and they work perfectly. We do not see caller ID from clients on either side. Our Grandstream phones say Eri and our XTen phones say Asterisk. We did a debug and I am pasting the output from both servers below. We tried setCallerId in several different ways. We see the value get passed to the IAX tunnel, but we do not see it in the call setup messages on the other side. Can someone shed any light on what we are doing wrong? Thanks, John == Spawn extension (local, 6201, 1) exited non-zero on '[EMAIL PROTECTED]/16384' -- Hungup '[EMAIL PROTECTED]/16384' -- Executing Dial(SIP/6201-dd24, IAX2/raleigh:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack -- Called raleigh:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 192.168.2.10 (format ULAW) -- Format for call is ULAW -- IAX2[arlington]/5 is ringing -- IAX2[arlington]/5 stopped sounds -- IAX2[arlington]/5 answered SIP/6201-dd24 = == Spawn extension (longdistance, 6201, 2) exited non-zero on 'SIP/-c4a6' -- Executing SetCallerID(SIP/7669-69b5, ) in new stack -- Executing Dial(SIP/7669-69b5, IAX2/arlington:[EMAIL PROTECTED]/[EMAIL PROTECTED] ocal) in new stack -- Called arlington:[EMAIL PROTECTED]/[EMAIL PROTECTED] -- Call accepted by 10.1.1.10 (format ULAW) -- Format for call is ULAW -- IAX2[raleigh]/3 is ringing -- IAX2[raleigh]/3 stopped sounds -- IAX2[raleigh]/3 answered SIP/7669-69b5 -- Hungup 'IAX2[raleigh]/3' ___ 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] problems with TDM400P
Wim, If ya don't need callerid then add the patch at http://www.nodomain.org/asterisk to zaptel and asterisk directories. I did this for UK callerid and the phone now rings on the first ring of the CO. Bit of a bodge but it works. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Wim Kerkhoff Sent: 02 June 2004 06:34 To: Asterisk-users Subject: [Asterisk-Users] problems with TDM400P Hi, We have two of these 4 port FXO cards. However, we are having some problems with incoming/outgoing calls. The latest version of Asterisk/zaptel from CVS is being used. Voicemail, internal SIP - SIP calls between Pingtel xpressa hard phones work terrific, echotest is fine, and so on. The zaptel and wcfxs modules load fine, and show all 8 FXO interfaces in dmesg: - Zapata Telephony Interface Registered on major 196 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO FXO Module 1: Installed -- AUTO FXO Module 2: Installed -- AUTO FXO Module 3: Installed -- AUTO FXO Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO FXO Module 1: Installed -- AUTO FXO Module 2: Installed -- AUTO FXO Module 3: Installed -- AUTO FXO Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) - Following problems have been observed, and are preventing us from dumping our existing Nortel Merdian PBX: 1. echo at beginning of call for several seconds, even with various combinations of echocancel and echotraining in zapata.conf 2. even though multiple incoming lines are connected, only the first ZAP channel is picking up. So if one line is in use, nobody else can call in even though there are other lines free. When in debug mode (-gcvvv) nothing is showing up that there's another call coming in. 3. channels don't always hang up properly - HookState shows as offhook for quite some time. 4. Asterisk Zap channels don't see an incoming call until 2 rings after the existing Nortel PBX sees it. Both people calling in and people answering don't like that. I've gone through whatever documentation and mailing list archives, but haven't been able to find working solutions. Have tried various combinations in zaptel.conf and zapta.conf but no luck yet :-( Ideas anyone? Thanks, Wim ___ 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] Caller ID with BT CD50
Cheers Tony. Your a star. Works a treat. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle Sent: 28 May 2004 00:48 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Caller ID with BT CD50 David J Carter wrote: Where would I find cdr-csv? Usually in /var/log/asterisk The line looks funny because of the line breaks. zapata.conf ukcallerid=yes callerid=asreceived signalling=fxs_ks channel = 1 : BT line channel = 2 : Telewest line I also have immediate=yes, but that shouldn't affect anything. Are you sure you've updated the modules correctly (done make/make install, done an rmmod on the old zaptel module and a modprobe on the new one)? There isn't much to go wrong beyond that... if you run asterisk with debugging you'll get a log if it finds a callerID but it's basically the same that goes into the cdr-csv file. Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ 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] Conference Server
I think if you use ztdummy that is all that is required. Un comment in the zaptel Makefile and recompile. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of pesb Sent: 27 May 2004 16:59 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Conference Server Hi there, I need to implement a SIP Conference Server. I've saw that asterisk has an application called meetme. But, it says that A ZAPTEL INTERFACE MUST BE INSTALLED FOR CONFERENCING FUNCTIONALITY. Is there any other way to implement a conference server without the need of having a ZAPTEL Interface? I need my conference server to work only with my SIP Phones. thanks in advance, Pablo Salinas ___ 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] Caller ID with BT CD50
Tony, I have downloades and installed the patches, (I think. I used patch -p0 /usr/src/zaptel/[patch], for bothe the zaptel ones, and [asterisk] for the asterisk one). I have addes the ukcallerid=yes to my zapata.conf, and also got callerid=asreceived set. The phones now ring without the screen showing starting simple switch 3-4 times, but alas no callerid on my GS phone. Any thoughts or hints appreciated. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle Sent: 26 May 2004 13:09 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Caller ID with BT CD50 I keep rolling buffer of the last couple of seconds of the incoming audio, so when the ring is detected the chan_zap driver can grab this and feed it to the callerid processing routines. If it's necessary to assign copyright to digium then there's no problem doing that. At the moment there's a rather lame 'ukcallerid=yes' command... it needs something better certainly but there's plenty of time to get that stuff right. The current patches are at http://www.nodomain.org/asterisk/ Ugh. V23 after first ring... It also matters of course if the cable co. has changed the wire data format - you might be able to grab the data but then not be able to make any sense of it.. Tony -- ___ 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] Caller ID with BT CD50
Where would I find cdr-csv? I have looked at all the asterisk directories. CLI is on all 3 lines I have into the house. I have addes the ukcallerid=yes to my zapata.conf, and also got callerid=asreceived set. The line looks funny because of the line breaks. zapata.conf ukcallerid=yes callerid=asreceived signalling=fxs_ks channel = 1 : BT line channel = 2 : Telewest line Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle Sent: 27 May 2004 22:31 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Caller ID with BT CD50 David J Carter wrote: I have addes the ukcallerid=yes to my zapata.conf, and also got callerid=asreceived set. No idea what that option does... The phones now ring without the screen showing starting simple switch 3-4 times, but alas no callerid on my GS phone. Check your cdr-csv file to see if asterisk is getting the incoming CID. It could just be the phone not displaying it. Presumably you have signed up with BT to have caller ID sent on your line? Tony -- Te audire no possum. Musa sapientum fixa est in aure. Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 ___ 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] Caller ID with BT CD50
Tony, Lost some of the mails on this topic somewhere. Does this need the BT50 mod or will the X100p now output the Caller ID? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Hoyle Sent: 26 May 2004 13:09 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Caller ID with BT CD50 Tim Robinson wrote: Tony - This sounds great. Are you monitoring the line constantly for the inbound caller ID or are you somehow detecting the polarity reversal? I keep rolling buffer of the last couple of seconds of the incoming audio, so when the ring is detected the chan_zap driver can grab this and feed it to the callerid processing routines. If it works and is stable, will you disclaim your code so that it will get merged into the main CVS? There should probably be a couple of settings in zapata.conf for the caller id coding scheme to be used for each card If it's necessary to assign copyright to digium then there's no problem doing that. At the moment there's a rather lame 'ukcallerid=yes' command... it needs something better certainly but there's plenty of time to get that stuff right. The current patches are at http://www.nodomain.org/asterisk/ since a lot of people here in UK have a line from BT and a cable co line, where the cable co either uses Bellcore after 1st ring, or V23 after 1st ring. So you need to be able to chose the method for each line. What a mess, eh? Ugh. V23 after first ring... It also matters of course if the cable co. has changed the wire data format - you might be able to grab the data but then not be able to make any sense of it.. Tony -- All your code belongs to Santa Tony Hoyle [EMAIL PROTECTED] Key ID: 104D/4F4B6917 2003-09-13 Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917 Phone(FWD): (0845 004 5566) 413300 ___ 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] How to share Zap channels in 2 Asterisk servers
Call the PBX300 using IAX2 from PBX200, make sure that the call goes into the context that allows dial out. Example. exten = _543219XX,1,StripMSD,5 exten = _9XX,2,Dial/[EMAIL PROTECTED]/BYEXTENSION The first line looks for an access code '54321' followed by the access code for an outside line '9' and then a number. You next strip the access code for IAX linking and pass the rest to the other Asterisk PBX. The Asterisk PBX then runs the exten as if on the local machine. Simple huh. There is most likely a simpler method, but this works for me. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 22 May 2004 16:40 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How to share Zap channels in 2 Asterisk servers Hello I am trying to setup Asterisk on 2 servers PBX300 and PBX200. PBX300 has X100P card with 1 telephone line. PBX200 don't have any Zap device. Softphone from PBX200 can talk to softphone on PBX300 but no outgoing call from PBX200. I can call from PBX300 outside but I am unable to configure soft Phone defined in PBX200 to dial out side using PBX300 Zap devices. I am geting error message Rejected connect attempt from PBX200. Please help if this is possible. Thanks Deepak This message was sent using IMP, the Internet Messaging Program. ___ 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] Making a SIP call
Check your sip.conf Make sure the dtmfmode is set the same as the phone. I had this before. Usually to dial an IP address you have a keystroke before you enter the address. I think on a Grandstream phone you press the menu button then the IP address. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: 21 May 2004 21:57 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Making a SIP call If someone could point me in the right direction I would much appreciate it. Here is my problem: My directions for my sip phone says to dial an ip address 12*34*65*78#. When I dial that into my phone my asterisk server is only picking up some of the numbers in the above example it would pick up 6578. Then of course not find it and ring busy on the phone. The same is true for dialing a regular phone number ( it seems to pick up 4 digits or so) I very new to setting this up so I imagine I need to make a change to a config file, but don't know where to start. ___ 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] Pots Extensions
Hi all,
I am either going daft or not reading things right.
I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I
have followed the examples for the conf files to the letter.
I can call the pots extensions OK from IAX clients, SIP clients and from the
incoming X100P cards.
But, if I pick up the handset to make a call all I get is the engaged tone
and the following message.
May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel 'ZAP/5-1'
sent into invalid extension 's' in context 'default' but no invalid handler.
If I am reading my configs then shouldn't they be going to the internal
context?
Do I need to set-up pots extensions somewhere like IAX Sip extensions?
=
zaptel.conf
fxsks=1-3
fxoks=4-7
loadzone=uk
zapata.conf
signalling=fxs_ks
context=incoming
channel = 1-3
signalling=fxo_ks
context=internal
channel = 4-7
extensions.conf
[internal]
exten = 4090,1,Dial,ZAP/4
exten = 4091,1,Dial,ZAP/5
exten = 4092,1,Dial,ZAP/6
exten = 4093,1,Dial,ZAP/7
exten = _9X.,Dial,ZAP/1,${EXTEN:1}
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RE: [Asterisk-Users] Pots Extensions
Lisa
Thanks for that, worked a treat.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Lisa Xie
Sent: 04 May 2004 17:33
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Pots Extensions
Did you put immediate=yes in your zapata.conf? I had similar problems
previously (I have T100p instead of X100p) and it is fixed when I put
immediate=no.
Lisa
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David J
Carter
Sent: Tuesday, May 04, 2004 12:43 PM
To: Asterisk User Group
Subject: [Asterisk-Users] Pots Extensions
Hi all,
I am either going daft or not reading things right.
I have just installed, (well 10 hours ago) a TDM40B and 3 X100P cards. I
have followed the examples for the conf files to the letter.
I can call the pots extensions OK from IAX clients, SIP clients and from
the
incoming X100P cards.
But, if I pick up the handset to make a call all I get is the engaged
tone
and the following message.
May 4 23:24:24 WARNING[221200]: pbx.c:1812 ast_pbx_run: Channel
'ZAP/5-1'
sent into invalid extension 's' in context 'default' but no invalid
handler.
If I am reading my configs then shouldn't they be going to the internal
context?
Do I need to set-up pots extensions somewhere like IAX Sip extensions?
=
zaptel.conf
fxsks=1-3
fxoks=4-7
loadzone=uk
zapata.conf
signalling=fxs_ks
context=incoming
channel = 1-3
signalling=fxo_ks
context=internal
channel = 4-7
extensions.conf
[internal]
exten = 4090,1,Dial,ZAP/4
exten = 4091,1,Dial,ZAP/5
exten = 4092,1,Dial,ZAP/6
exten = 4093,1,Dial,ZAP/7
exten = _9X.,Dial,ZAP/1,${EXTEN:1}
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RE: Caller ID Re: [Asterisk-Users] Re: Support Digium
Mark J Elkins wrote Um - Digium wants you to buy their hardware - but there is a CLID issue.. would it not make more financial sense to insert a dumb ISDN card (or two), and upgrade your PSTN to ISDN??? Would this not assist Digium in making sure CLID worked in the UK??? Isn't this a bit like cutting of the nose to spite the face. UK PSTN lines costs £30 /Qtr UK ISDN costs £65 /qtr, you could buy two X100P's every year and still be in pocket by staying with PSTN. There was a post on the list in the not to distant past where someone had written two small scripts for getting the information from a BT50 and a serial modification and passing it to asterisk. Still seems the best way in the interim. As has been said many times in the list Digium have given us this software, we don't have to give them a hard time in return. Not a fair payback. Dave ___ 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] smallest phone
Just tried Pulver but it's in a password protected area. Any idea of the other places? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jason Williams Sent: 25 April 2004 10:31 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] smallest phone There are later versions of the firmware that work better Available from pulver and other places Jason At 03:07 25/04/2004 -0500, you wrote: I do have a WISIP and it doesnt give me any problems im all day long on the street using it. You cant talk of a phone you havent even touch Miguel On Fri, 2004-04-23 at 10:33, Andrew Kohlsmith wrote: why not wisip? its size its like a regular cellphone and it uses wifi Because it sucks ass? Check the archives for some very valid gripes about the device. -A. ___ 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 ___ 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] inbound calls better quality than outbound calls on X100P
I have my RX at 4.0 ant TX at 8.0, I get slight echo for the first 5-6 seconds then all OK. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Stenton Sent: 22 April 2004 17:07 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] inbound calls better quality than outbound calls on X100P I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is wrong, I'm using the latest zaptel and asterisk from the cvs head as of today. Chris ___ 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] Asterisk + GrandStream SIP phones
What does your extensions.conf look like? Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of pesb Sent: 29 March 2004 18:48 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk + GrandStream SIP phones -This is my 'sip.conf' file: ;* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secret= reinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1004 nat=1 disallow=all allow=ulaw allow=alaw [1005] type=friend username=1005 secret= reinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1005 nat=1 disallow=all allow=ulaw allow=alaw ;*** -And this is the basic seting of my two GrandStream SIP phones: ***[1005] IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:empty SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***[1004] IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:empty SIP User ID:1004 Authenticate ID:1004 Authenticate Password:123 Name:1004 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ** I have 2 SIP GrandStream phones, both phones are correctly registered to the Asterisk server. But, when I try to make a call from registered phone '1005' to registered phone '1004', dialing 1004, Asterisk responds with the 'Status: 404 Not Found' message. How do I have to dial? What else do I need to set? Find attached my traffic captured on ethereal. ___ 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 + GrandStream SIP phones
Try this small extensions.conf Don't think I have missed owt. My config files are here, you just need to add your own extension numbers. http://www.codepipe.com/id25.htm Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of pesb Sent: 29 March 2004 19:26 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk + GrandStream SIP phones -This is my 'sip.conf' file: ;* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secret= reinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1004 nat=1 disallow=all allow=ulaw allow=alaw [1005] type=friend username=1005 secret= reinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1005 nat=1 disallow=all allow=ulaw allow=alaw ;*** -And this is the basic seting of my two GrandStream SIP phones: ***[1005] IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:empty SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***[1004] IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:empty SIP User ID:1004 Authenticate ID:1004 Authenticate Password:123 Name:1004 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ** I have 2 SIP GrandStream phones, both phones are correctly registered to the Asterisk server. But, when I try to make a call from registered phone '1005' to registered phone '1004', dialing 1004, Asterisk responds with the 'Status: 404 Not Found' message. How do I have to dial? What else do I need to set? Find attached my traffic captured on ethereal. extensions.conf Description: Binary data
RE: [Asterisk-Users] Semi OT: WiSIP and WEP
Hi Gavin, Works OK with my 128-Bit WAP. Remove the Space or put in an underscore and try again. Regards Dave -Original Message- Gavin Adams wrote: - Received my Pulver WiSIP phone a couple days ago. Has anyone successfully gotten the phone to work with 128-bit WEP? I've tried entering the key via the keyboard (ugh), turning off WEP then adding the key via the web browser (minor ugh), and all steps in between. The only thing that may be an issue is that my SSID has a space in it Test WAP. When I view it the first time on the phone, it appears correctly. However, the second time, only the first word appears Test. Promising phone if I can ever get it to work on my network. Regards, --- Gavin Adams Promisant (Technology) Ltd. Atlanta, GA ___ 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: Asterisk SIP + Grandstream 100 + sip.conf phone HELP
I use GS 101 102, have a look at my configs at http://www.codepipe.com/id25.htm . Hope they help. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Stephen R. Besch Sent: 23 March 2004 20:22 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk SIP + Grandstream 100 + sip.conf phone HELP --snip-- I am having trouble setting the /etc/asterisk/sip.conf file. This is my file: 1) Add in the [general] section: disallow=all allow=ulaw allow=alaw allow=any other codec that you want to (or can) support. While some have found that this must be specified for each and every phone, I have found that it works fine specified just once in the general section. [243075] type = friend context = default secret = gol host = dynamic callerid = fono75 243075 2) Include dtfmmode=info or inband and match to phone's setting 3) I may have been too tired at the time, but once I tried using long extensions (more than 5 digits) and could not make them work either - same error you are getting. I would limit your extensions to 4 digits and see if it helps. 4) You may also need to add canreinvite=no to each phone definition. and our SIP phones configuration are the following: SIP Server: 192.168.0.102 Outbound Proxy: Empty 5) I would set this to be the same as the server if you want to make outbound calls. Hope this helps Stephen R. Besch ___ 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] EM Signalling
Hi all, I may need to connect to a system with EM connectivity. Am I right in assuming a T1 card and Channel Bank will give me this connectivity? Regards Dave ___ 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] EM Signalling
Thanks for all the replies. The system I may have to connect to has some spare analogue 2/4 wire EM trunks. I have two Multitech units but am having some difficulty with the H323 on asterisk at the moment. I think I have built everything OK but get problems starting asterisk with the h323 modules loaded. any help on the h323 would be appreciated as well. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of George Pajari Sent: 22 March 2004 22:28 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] EM Signalling On Monday 22 March 2004 15:35, David J Carter wrote: I may need to connect to a system with EM connectivity. Am I right in assuming a T1 card and Channel Bank will give me this connectivity? Perhaps we ought to make sure we're talking about the same thing. Mr. Carter: are you talking about EM signalling on digital trunks or four-wire analog EM trunks? g. ___ 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] UK BT caller ID revisted
John Lawrence wrote Hi all, Does anyone know the procedure for adding a serial output to a cheap caller display unit. If I can find a way of doing this then I'm sure there will be away for linux to take the CallerID info, write it to a file, * to open that file an read the number from it. TIA Jon I am going to a Radio Rally tommorow and I will buy a couple of stand alone Caller Display units to strip and get a serial output from them. Watch this space. Regards Dave ___ 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] Can i do voice chat without using the hardware
My aim is that, i want to connect my PC (where i installed the asterisk) to another PC in my network for voice chating. For this purpose, what are the steps to be done? which are the files to be modified. I would like to make use of the existing Hardware (sound card, network card etc), i am not using any extra hardware. Is X-Lite work in Linux? or any compatible s/w that works under linux? Have a look at these sites: - http://www.codepipe.com/id25.htm http://www.jaredsmith.net/misc/hgta/ http://www.wwworks-inc.com/asterisk/ http://www.fnords.org/~eric/asterisk/ http://bcwireless.net/moin.cgi/VoIPHowTo http://www.automated.it/guidetoasterisk.htm http://www.asterisk.org/index.php?menu=support http://www.voip-info.org/wiki-Asterisk+config+files http://www.voip-info.org/tiki-index.php?page=Asterisk If you have the CLI prompt then your almost there. If you have the audio set up in asterisk then you can use a headset/microphone to call the other party. CLIdial 1234 when finished CLIhangup Simple huh? Regards Dave ___ 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] Newbie Start Question
Just one question, Why do you want users sent to the Demo at Digium? take a look at: - http://www.codepipe.com/id25.htmI have some sample files there. If you want to contact me off list [EMAIL PROTECTED] the we will not tie the list up with 8000 posts for every reply. Regards Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Mamadou Lamine KASent: 19 March 2004 14:30To: [EMAIL PROTECTED]Subject: [Asterisk-Users] Newbie Start Question Hi Everybody, I am very new to Asterisk. I want to set up a PBX and an IVR server with it. I have a wildcard X100P and a TDM400P on my RedHat box. I have installed Asterisk and the devices and everything seems OK. (Asterisk Ready) Now I want to launch the Demo context in /etc/asterisk/extensions.conf so that when a call comes it is directed on that context. How shall I proceed? I have of course read the Asteriskhandbook but it is too the theorical to me. Could someone tell me where i can find exact informations on how to set up and how to use IVR server with Asterisk. Any help will be highly appreciated. Thanks in advance Mamadou Lamine KA
RE: [Asterisk-Users] x100p CLI in the UK
Chris, May be a bad card, or more likely Microfilter, I have had mine on the same line as the ADSL for 3 months now and no problems. As for UK CLI I will be glad when I can get CLI from either BT or Telewest. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Lee Sent: 16 March 2004 00:26 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] x100p CLI in the UK First, is the lack of UK CLI on the x100P hardware or software related? Secondly, My US Robotics Voice modem does get UK CLI, so could I get UK CLI and the same functionality as the x100p using a USR Modem with *? Has anyone done this? As an aside, has anyone experienced or solved the problem with the x100p producing a loop condition on the PSTN line (it really mucks up my ADSL connection something horrid when it is connected). I think it is due to an impedance mismatch between the card and the network, but have no way of testing these things. (Dont know enough to just get out my meter and start probing without risk of killing my x100p or the POTS Line) I know the Loop condition is there as a kindly BT eng was monitoring the line and asking me to plug things in, when the x100p was plugged in he said something along the lines of: theres your problem, what did you just plug in? It is creating a 36 K Ohm Loop condition Now the router is not the most stable at the best of times but plug in the x100p and the line bounces up and down like there is no tomorrow. Regards Chris. ___ 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] Help on two subjects
Hi All, I have now got my '*' server up and running quite good. As stated in earlier posts I am no Linux guru, so a bit of hand holding required. First Subject. I would now like to add h323 boxes to the '*' server, I have looked through the wiki and followed the instructions about what I need but I am a little thick as I can't seem to get to grips with it. Has anybody got a dummies step by step guide to installing things needed for h323. ala 1. turn on your server. 2. log onto your server. 3. make a cup of coffee because ya gonna need it. 4. .. and so on. Second Subject. I have never used or seen a channel bank, but I think it is what I require for a project I am looking at. I have 12 Analogue (CO) lines that I would like to bring into the '*' server. I have 12 Analogue POTS that I would like to connect to the '*' server, these are along with SIP phones (Grandstream), and IAX clients. The later two I have no problems with, see First Subject for the other failings. If any one can help then please either answer on or off list. Regards thanks in advance. Dave ___ 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] X100P and TDM400 questions
hi,
Try
exten = _9.,1,Dial(Zap/1/${EXTEN:1})
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of randulo
Sent: 12 March 2004 14:54
To: asterisk list
Subject: [Asterisk-Users] X100P and TDM400 questions
I have the dev kit installed and the X100P answers calls and * routes
them as expected. I am not able to dial out at all:
[analog-out]
exten = _9.,1,Dial(Zap/1/$EXTEN:1)
exten = _9.,2,Congestion
included up in the default section
shouldn't this take any call beginning with 9, strip the 9 and dial it?
I happen to be in France; does that matter? Since all US modems work
here, I assumed the X100 would too. Would it be waiting for a dial tome
that never comes?
TDM
I can not see the TDM400 in /proc/interrupt - shouldn't there be an
entry for it? I've tried every PCI slot but it doesn't seem to be seen
in any of them.
thx for any comments
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RE: [Asterisk-Users] X100P dial in/out to sip phones
Simon,
Caller ID does not work in the UK, well not on my BT or Telewest line's.
Have a look at my sample configs http://www.codepipe.com/id25.htm , I am
also in the UK and these work for me.
Give me a call if ya want to chat about it.
Regards
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Simon
Chappell
Sent: 07 March 2004 16:46
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] X100P dial in/out to sip phones
Hello all
I have recently stumbled accross voip and asterisk.
We have a small network of vpns running in the uk. I have managed to get
the sip phones dialing each other through asterisk and it is working
great. (we are having long free conversations and that is something to
get excited about)..
My problem is that I cannot get the X100P i recently bought to dial out
or do anything with incoming calls.
I did loads of googling and found this snippet that made the zaptel card
moan at me about callerid ask me to type a number then do nothing but
offer silence..
[inbound-analog]
exten = s,1,Zapateller(answer|nocallerid)
exten = s,2,NoOp
exten = s,2,Macro(record-on,${PHONE1},${CALLERIDNUM})
exten = s,3,PrivacyManager
exten = s,4,Dial(${PHONE1},15,Ttm)
exten = s,5,Answer
exten = s,6,Wait(1)
exten = s,7,Playback(new/hello)
exten = s,8,Playback(new/marisa-john-not-in-momnt)
exten = s,9,Playback(new/theyre-rattlesnake-wrstling)
exten = s,10,Voicemail(u${PHONE1VM})
exten = s,11,Hangup
exten = s,108,Wait(2)
exten = s,109,Voicemail(b${PHONE1VM})
exten = s,110,Hangup
If i rem out that and run asterisk with -vvg i get this when i dial in
to the x100p
Mar 7 16:43:41 NOTICE[245776]: chan_zap.c:4624 ss_thread: Got event 2
(Ring/Answered)...
Mar 7 16:43:44 NOTICE[245776]: chan_zap.c:4624 ss_thread: Got event 2
(Ring/Answered)...
Mar 7 16:43:44 NOTICE[245776]: chan_zap.c:4624 ss_thread: Got event 2
(Ring/Answered)...
Mar 7 16:43:49 WARNING[245776]: chan_zap.c:4695 ss_thread: CallerID
returned with error on channel 'Zap/1-1'
Mar 7 16:43:49 WARNING[245776]: pbx.c:1778 ast_pbx_run: Channel
'Zap/1-1' sent into invalid extension 's' in context 'default', but no
invalid handler
So i feel i am getting there..
I would like the extensions to dial out and ring when the line rings..
can anyone give me a clue or point me in the right direction
I am in the UK by the way if that makes a difference.
Many thanks in advance
Simon
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RE: [Asterisk-Users] Re: Grandstream Budgetone SIP registration fails
Tony, Have a look here http://www.codepipe.com/id25.htm these are my working examples. I have 6 GS phones. The GS set-up's are from extersion 8002 onwards in sip.conf. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tony Mountifield Sent: 06 March 2004 21:04 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Grandstream Budgetone SIP registration fails In article [EMAIL PROTECTED], Jean-Marc V. Liotier [EMAIL PROTECTED] wrote: Someone on the list certainly has a working setup with Asterisk and Grandstream Budgetone phones, I would be grateful if their SIP configuration was posted to the list. Quite unexpectedly I found no complete example of such working setup on the Web, maybe because it was so simple that no one thought that posting it would be useful to anyone. One I get mine working I shall post the parameters ! Well my sip.conf looks like this: -- ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context=from-sip-external ; send unknown SIP callers to this context allow=ulaw allow=ilbc ; ; Tony's phone ; [2000] type=friend username=2000 secret=password host=dynamic context=from-sip-internal mailbox=2000 callerid=2000 dtmfmode=info ; ; Rachel's phone ; [2001] type=friend username=2001 secret=password host=dynamic context=from-sip-internal mailbox=2001 callerid=2001 dtmfmode=info -- Then in the admin interface for Tony's phone I have the following: IP address: dynamic from DHCP SIP server: IP of Asterisk server Outbound proxy: empty SIP User ID: 2000 Authenticate ID: 2000 Auth password: password Vocoder choices (in order): PCMU, PCMA, then others SIP user ID is phone number: Yes SIP Registration: Yes Clear reg on reboot: No Reg expiration: 3 Early dial: No Local SIP port: 5060 Local RTP port: 5004 Use random port: No NAT Traversal: No Send DTMF: Via SIP INFO I think that's all the likely relevant ones. Hope this helps Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.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 ___ 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] on hold music from a mp3 stream or sound card input?
No need to string them together. Just put them in the MP3 directory and it will play them one by one, taht's all i have done. My largest MP3 plays for 20 minutes. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Dean Collins Sent: 04 March 2004 07:05 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] on hold music from a mp3 stream or sound card input? Can I ask an addendum question to this. How large can the mp3 file be? I haven't played with this yet but wondering if I can connect about 20-30 mp3's together so my people on hold don't hear the same music very often. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Thursday, 4 March 2004 12:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] on hold music from a mp3 stream or sound card input? Hi Folks, Rather than have my hold music play from a sound file I'd like to have a live feed from a sound card input or MP3 stream. Is this doable and if so how? -- Mark Phillips, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Anybody managed to call a phone through sipgate.de
Hi,
Are you behind a NAT/Firewall?
dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 28 February 2004 11:04
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
David Hajek wrote:
| Is there english version of their sipgate.de website?
no ... I just tried the google translater - it did not work (for me) I
think the translation programs don't work with php pages...
Birk
|
| -D
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Birk Bremer
|Sent: Friday, February 27, 2004 7:06 PM
|To: [EMAIL PROTECTED]
|Subject: Re: [Asterisk-Users] Anybody managed to call a phone
|through sipgate.de
|
| Hi David,
|
| no the number after the slash is necessary (and yes this is
| my number) Without that slash/number I'm not able to get a
| call anymore.
|
| But thanks
|
| Birk
|
|
|
|
| David J Carter wrote:
| | Hi,
| |
| | I would be tempted to get rid of the slash and number on
| the register
| line,
| | unless your asterisk extension is 02115800.
| |
| | dave
| |
| | -Original Message-
| | From: [EMAIL PROTECTED]
| | [mailto:[EMAIL PROTECTED] Behalf Of
| Birk Bremer
| | Sent: 27 February 2004 16:47
| | To: [EMAIL PROTECTED]
| | Subject: [Asterisk-Users] Anybody managed to call a phone through
| | sipgate.de
| |
| |
| | Hello everybody,
| |
| | has anybody managed to call a (old fashioned) phone using
| Sipgate.de
| | and asterisk? (yes I have money on my account :-) )
| |
| |
| | The configuration I got from the sipgate.de people is at
| the botton of
| | the mail
| |
| |
| | Here is mine:
| |
| | sip.conf:
| |
| | register = 800:[EMAIL PROTECTED]/02115800
| |
| | [sipgate]
| | type=friend
| | username=800
| | secret=SECRET
| | host=sipgate.de
| | fromuser=800
| | fromdomain=sipgate.net
| | nat=no
| | ;dtmfband=3Dinband
| | context=sipin
| | canreinvite=no
| |
| |
| | extension.conf:
| | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| |
| | To be called on my sipgate number - no problem
| |
| | If I want to call somebody I get the following error:
| |
| | When I call a number directly out of the softphone:
| | Executing Dial([EMAIL PROTECTED]/2,
| SIP/[EMAIL PROTECTED]|30|tr)
| | in new stack
| | ~-- Called [EMAIL PROTECTED]
| | ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| | ~ == No one is available to answer at this time
| | ~-- Hungup '[EMAIL PROTECTED]/2
| |
| |
| |
| | when I use the webinterface at sipgate.de I get a ring at my
| | softphone, when I pick the call I get the message (in the appearing
| | box) Teilnehmer nicht gefunden - User/Number not found
| |
| | sometimes (while tried different config. I also got (at *
| console) to
| | many hops...
| |
| |
| | Has anybody managed this - can you please send me your
| configuration
| | (sip, extensions) or can anybody help
| |
| | Thanks in advance
| |
| | Birk Bremer
| |
| |
| |
| |
| |
| | The configuration the sipgate people suggest:
| |
| | ~ register = 800:[EMAIL PROTECTED]/800
| | ^ can't be correct
| | |
| | |
| | |
| | | [sipgate]
| | |
| | | type=friend
| | |
| | | username=800
| | |
| | | secret=sipgatepasswort
| | |
| | | host=sipgate.de
| | |
| | | fromuser=800
| | |
| | | fromdomain=sipgate.net
| | |
| | | nat=yes
| | |
| | | ;dtmfband=inband
| | |
| | | context=incomingsipgate
| | |
| | | canreinvite=no
| | |
| | |
| | |
| | | Aus der extensions.conf :
| | |
| | |
| | |
| | | [incomingsipgate]
| | |
| | | exten = h,1,Hangup
| | |
| | | exten = 800,1,Dial(SIP/internestelefon,20,tr)
| | |
| | |
| | |
| | | [sipgate]
| | |
| | | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| | |
| | | exten = _9.,2,Playback(invalid)
| | |
| | | exten = _9.,3,Hangup
|
| ___
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| http://lists.digium.com/mailman/listinfo/asterisk-users
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| ~ http://lists.digium.com/mailman/listinfo/asterisk-users
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RE: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
Hi again,
What is your sipgate number, I have just setup my asterisk to call a sipgate
numbar and it rings.
If you want to call me, then try my IAXTEL # 1 700 818 8820
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 28 February 2004 11:04
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
David Hajek wrote:
| Is there english version of their sipgate.de website?
no ... I just tried the google translater - it did not work (for me) I
think the translation programs don't work with php pages...
Birk
|
| -D
|
|
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Birk Bremer
|Sent: Friday, February 27, 2004 7:06 PM
|To: [EMAIL PROTECTED]
|Subject: Re: [Asterisk-Users] Anybody managed to call a phone
|through sipgate.de
|
| Hi David,
|
| no the number after the slash is necessary (and yes this is
| my number) Without that slash/number I'm not able to get a
| call anymore.
|
| But thanks
|
| Birk
|
|
|
|
| David J Carter wrote:
| | Hi,
| |
| | I would be tempted to get rid of the slash and number on
| the register
| line,
| | unless your asterisk extension is 02115800.
| |
| | dave
| |
| | -Original Message-
| | From: [EMAIL PROTECTED]
| | [mailto:[EMAIL PROTECTED] Behalf Of
| Birk Bremer
| | Sent: 27 February 2004 16:47
| | To: [EMAIL PROTECTED]
| | Subject: [Asterisk-Users] Anybody managed to call a phone through
| | sipgate.de
| |
| |
| | Hello everybody,
| |
| | has anybody managed to call a (old fashioned) phone using
| Sipgate.de
| | and asterisk? (yes I have money on my account :-) )
| |
| |
| | The configuration I got from the sipgate.de people is at
| the botton of
| | the mail
| |
| |
| | Here is mine:
| |
| | sip.conf:
| |
| | register = 800:[EMAIL PROTECTED]/02115800
| |
| | [sipgate]
| | type=friend
| | username=800
| | secret=SECRET
| | host=sipgate.de
| | fromuser=800
| | fromdomain=sipgate.net
| | nat=no
| | ;dtmfband=3Dinband
| | context=sipin
| | canreinvite=no
| |
| |
| | extension.conf:
| | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| |
| | To be called on my sipgate number - no problem
| |
| | If I want to call somebody I get the following error:
| |
| | When I call a number directly out of the softphone:
| | Executing Dial([EMAIL PROTECTED]/2,
| SIP/[EMAIL PROTECTED]|30|tr)
| | in new stack
| | ~-- Called [EMAIL PROTECTED]
| | ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| | ~ == No one is available to answer at this time
| | ~-- Hungup '[EMAIL PROTECTED]/2
| |
| |
| |
| | when I use the webinterface at sipgate.de I get a ring at my
| | softphone, when I pick the call I get the message (in the appearing
| | box) Teilnehmer nicht gefunden - User/Number not found
| |
| | sometimes (while tried different config. I also got (at *
| console) to
| | many hops...
| |
| |
| | Has anybody managed this - can you please send me your
| configuration
| | (sip, extensions) or can anybody help
| |
| | Thanks in advance
| |
| | Birk Bremer
| |
| |
| |
| |
| |
| | The configuration the sipgate people suggest:
| |
| | ~ register = 800:[EMAIL PROTECTED]/800
| | ^ can't be correct
| | |
| | |
| | |
| | | [sipgate]
| | |
| | | type=friend
| | |
| | | username=800
| | |
| | | secret=sipgatepasswort
| | |
| | | host=sipgate.de
| | |
| | | fromuser=800
| | |
| | | fromdomain=sipgate.net
| | |
| | | nat=yes
| | |
| | | ;dtmfband=inband
| | |
| | | context=incomingsipgate
| | |
| | | canreinvite=no
| | |
| | |
| | |
| | | Aus der extensions.conf :
| | |
| | |
| | |
| | | [incomingsipgate]
| | |
| | | exten = h,1,Hangup
| | |
| | | exten = 800,1,Dial(SIP/internestelefon,20,tr)
| | |
| | |
| | |
| | | [sipgate]
| | |
| | | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| | |
| | | exten = _9.,2,Playback(invalid)
| | |
| | | exten = _9.,3,Hangup
|
| ___
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| http://lists.digium.com/mailman/listinfo/asterisk-users
| To UNSUBSCRIBE or update options visit:
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RE: [Asterisk-Users] Anybody managed to call a phone through sipgate.de
Birk,
Even using VPN to get to the server you will still have I assume a private
IP address on the VPN side. This will pass through a NAT/Firewall to the
outside world. This may or may not be on the server you connect to, but I
would bet you still pass through a NAT/Firewall.
I assume your connection is something like: -
Softphone Asterisk VPN to Server -- Server ---
Firewall/NAT/Router - Internet
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 28 February 2004 11:32
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
The Server I use is somewhere in the Internet with a real ip. Myself and
others connect to the server via vpn in order to go through various
firewalls. Since I can get calls but only can't place calls (via
sipgate.de) I don't think it is a firewall matter...
Birk
David J Carter wrote:
| Hi,
|
| Are you behind a NAT/Firewall?
|
| dave
|
| -Original Message-
| From: [EMAIL PROTECTED]
| [mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
| Sent: 28 February 2004 11:04
| To: [EMAIL PROTECTED]
| Subject: Re: [Asterisk-Users] Anybody managed to call a phone through
| sipgate.de
|
|
| David Hajek wrote:
| | Is there english version of their sipgate.de website?
|
|
| no ... I just tried the google translater - it did not work (for me) I
| think the translation programs don't work with php pages...
|
| Birk
|
|
| |
| | -D
| |
| |
| |-Original Message-
| |From: [EMAIL PROTECTED]
| |[mailto:[EMAIL PROTECTED] On Behalf Of
| |Birk Bremer
| |Sent: Friday, February 27, 2004 7:06 PM
| |To: [EMAIL PROTECTED]
| |Subject: Re: [Asterisk-Users] Anybody managed to call a phone
| |through sipgate.de
| |
| | Hi David,
| |
| | no the number after the slash is necessary (and yes this is
| | my number) Without that slash/number I'm not able to get a
| | call anymore.
| |
| | But thanks
| |
| | Birk
| |
| |
| |
| |
| | David J Carter wrote:
| | | Hi,
| | |
| | | I would be tempted to get rid of the slash and number on
| | the register
| | line,
| | | unless your asterisk extension is 02115800.
| | |
| | | dave
| | |
| | | -Original Message-
| | | From: [EMAIL PROTECTED]
| | | [mailto:[EMAIL PROTECTED] Behalf Of
| | Birk Bremer
| | | Sent: 27 February 2004 16:47
| | | To: [EMAIL PROTECTED]
| | | Subject: [Asterisk-Users] Anybody managed to call a phone through
| | | sipgate.de
| | |
| | |
| | | Hello everybody,
| | |
| | | has anybody managed to call a (old fashioned) phone using
| | Sipgate.de
| | | and asterisk? (yes I have money on my account :-) )
| | |
| | |
| | | The configuration I got from the sipgate.de people is at
| | the botton of
| | | the mail
| | |
| | |
| | | Here is mine:
| | |
| | | sip.conf:
| | |
| | | register = 800:[EMAIL PROTECTED]/02115800
| | |
| | | [sipgate]
| | | type=friend
| | | username=800
| | | secret=SECRET
| | | host=sipgate.de
| | | fromuser=800
| | | fromdomain=sipgate.net
| | | nat=no
| | | ;dtmfband=3Dinband
| | | context=sipin
| | | canreinvite=no
| | |
| | |
| | | extension.conf:
| | | exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
| | |
| | | To be called on my sipgate number - no problem
| | |
| | | If I want to call somebody I get the following error:
| | |
| | | When I call a number directly out of the softphone:
| | | Executing Dial([EMAIL PROTECTED]/2,
| | SIP/[EMAIL PROTECTED]|30|tr)
| | | in new stack
| | | ~-- Called [EMAIL PROTECTED]
| | | ~-- Got SIP response 403 Forbidden back from 217.10.79.9
| | | ~ == No one is available to answer at this time
| | | ~-- Hungup '[EMAIL PROTECTED]/2
| | |
| | |
| | |
| | | when I use the webinterface at sipgate.de I get a ring at my
| | | softphone, when I pick the call I get the message (in the appearing
| | | box) Teilnehmer nicht gefunden - User/Number not found
| | |
| | | sometimes (while tried different config. I also got (at *
| | console) to
| | | many hops...
| | |
| | |
| | | Has anybody managed this - can you please send me your
| | configuration
| | | (sip, extensions) or can anybody help
| | |
| | | Thanks in advance
| | |
| | | Birk Bremer
| | |
|
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Version: GnuPG v1.2.4 (GNU/Linux)
Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
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n54rHyhWAMcQSCKXZNTbEfk=
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RE: [Asterisk-Users] wisip firmware, updates, features??
Hi Johnathan, I wouldn't mind a copy of the firmware if you could send it. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jonathan Moore Sent: 28 February 2004 19:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] wisip firmware, updates, features?? NOt sure if there is an official download site, but I just recieved a copy of the updated firmware from pulver. I can send it to you if you like. I have emailed back asking for instructions on how to load. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Miguel Cavazos [EMAIL PROTECTED]: hi guys finally i got my wisip this week and im very happy with it. It works but i was wondering anyone know where can i find new firmware, updates or a wish list? I cross emails with jeff pulver about having a small http browser for auth on starbucks hotspots mcdonalds or prodigy movil(mexico). Even to check some text things via web maybe email??? He seems not to be so intrested so ill try emailing the manufacture. However if someone has a useful url or can tell me where to find this information please send me an email. Miguel ___ 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 Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] Anybody managed to call a phone through sipgate.de
Hi,
I would be tempted to get rid of the slash and number on the register line,
unless your asterisk extension is 02115800.
dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Birk Bremer
Sent: 27 February 2004 16:47
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Anybody managed to call a phone through
sipgate.de
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello everybody,
has anybody managed to call a (old fashioned) phone using Sipgate.de and
asterisk? (yes I have money on my account :-) )
The configuration I got from the sipgate.de people is at the botton of
the mail
Here is mine:
sip.conf:
register = 800:[EMAIL PROTECTED]/02115800
[sipgate]
type=friend
username=800
secret=SECRET
host=sipgate.de
fromuser=800
fromdomain=sipgate.net
nat=no
;dtmfband=3Dinband
context=sipin
canreinvite=no
extension.conf:
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
To be called on my sipgate number - no problem
If I want to call somebody I get the following error:
When I call a number directly out of the softphone:
Executing Dial([EMAIL PROTECTED]/2, SIP/[EMAIL PROTECTED]|30|tr)
in new stack
~-- Called [EMAIL PROTECTED]
~-- Got SIP response 403 Forbidden back from 217.10.79.9
~ == No one is available to answer at this time
~-- Hungup '[EMAIL PROTECTED]/2
when I use the webinterface at sipgate.de I get a ring at my softphone,
when I pick the call I get the message (in the appearing box)
Teilnehmer nicht gefunden - User/Number not found
sometimes (while tried different config. I also got (at * console) to
many hops...
Has anybody managed this - can you please send me your configuration
(sip, extensions) or can anybody help
Thanks in advance
Birk Bremer
The configuration the sipgate people suggest:
~ register = 800:[EMAIL PROTECTED]/800
^ can't be correct
|
|
|
| [sipgate]
|
| type=friend
|
| username=800
|
| secret=sipgatepasswort
|
| host=sipgate.de
|
| fromuser=800
|
| fromdomain=sipgate.net
|
| nat=yes
|
| ;dtmfband=inband
|
| context=incomingsipgate
|
| canreinvite=no
|
|
|
| Aus der extensions.conf :
|
|
|
| [incomingsipgate]
|
| exten = h,1,Hangup
|
| exten = 800,1,Dial(SIP/internestelefon,20,tr)
|
|
|
| [sipgate]
|
| exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],30,tr)
|
| exten = _9.,2,Playback(invalid)
|
| exten = _9.,3,Hangup
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Comment: Using GnuPG with Debian - http://enigmail.mozdev.org
iD8DBQFAP3R87QhrwFQeHVsRAjx+AJ9SvPdV4YY5iSZflo9XX/Xi97YM3wCghniD
5HUMSd5i2HUik75eajuJtpU=
=01sy
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[Asterisk-Users] Off topic question
Hi, Sorry for the of topic question, but where else do you get so many telco guys in one place. I have a customer who is moving to Australia and was on ADSL here in the UK. Q) Is ADSL a standard? and will his router/modem work in AU? I have told him a tentative yes but would page the oracles for clarification. Regards Dave ___ 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] Unable to create channel of type 'Zap'
I had this after my last CVS update.
A line in Zaptel.conf was set to fxsls=1 instaead of fxsks=1
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Wim Venneman
Sent: 24 February 2004 19:17
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'
Thanks Derek,
Changed the channel = 1 to channel = 1, makes no difference.
Wim
- Original Message -
From: Derek Samford [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Tuesday, February 24, 2004 6:38 PM
Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
Wim,
Made one more change below in Zapata.conf
It should be channel = 1
-Original Message-
From: Wim Venneman [mailto:[EMAIL PROTECTED]
Sent: Monday, February 23, 2004 4:46 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Unable to create channem of type 'Zap'
Thanks for the help !
Made changes, still the same message.
I have two NIC's with IRQ 11
The FXO card has IRQ10 (and no other card has IRQ10)
Wim
- Original Message -
From: Brent Franks [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 23, 2004 10:21 PM
Subject: RE: [Asterisk-Users] Unable to create channem of type 'Zap'
Wim, I made some changes to your Zapata.conf and zaptel.conf config
files below.
Hope this helps.
Also, do a less /proc/interrupts and see if the card is on it's own
IRQ.
- Brent
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wim
Venneman
Sent: Monday, February 23, 2004 3:10 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Unable to create channem of type 'Zap'
Can anyone help me, (after a two day search, also on the mailing list)
I have the following situation:
Asterisk works fine, until I added a FXO card. (Digium)
When I tried to call to the pstn I have the following error
Executing Dial(SIP/Phone2-fc49, Zap/1/2355) in new stack
NOTIVE[16401]: FILE APP_DIAL.C, LINE 516 (DIAL_EXEC): UNABLE TO CREATE
CHANNEL OF TYPE 'ZAP'
== Everyone is busy at this time
When I start Asterisk I have no error
Only the following isn't right:
ZAP SHOW CHANNELS = No channels
modprobe wcfxo = ok (no errors)
I have following config.
ZAPATA
[channels]
language=en
group=1
pickupgroup=1
context=incoming
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
immediate=yes
musiconhold=default
channel = 1
ZAPTEL
loadzone = us
defaultzone = us
fxsks = 1
EXTENSION
[incoming]
exten = s,1,Dial(SIP/Phone1SIP/Phone3,20,tr)
[outgoing]
exten = _0X.,1,Dial,Zap/1/${EXTEN:1}
IN [SIP]
include = outgoing
I'm don't know what I can change to the config.
Anyone an idea
Thanks,
Wim
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RE: [Asterisk-Users] Budgetone phones from FWD
I ordered a WiSIP from them on Friday last, and had confirmation yesterday the it was in Transit from the US to The UK. E-Mail them they are very good at responding. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jonathan Moore Sent: 18 February 2004 18:21 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Budgetone phones from FWD There is an email contact in the order form area. I have had very good luck with emailing her and having questions answered about availability. -- Jonathan Moore Director of Technology Winfield Public Schools Office 620.221.5100 Fax 620.221.0508 Quoting Jason T. Nelson [EMAIL PROTECTED]: I recently ordered a few phones from them for some testing I'm doing, but the charges still haven't appeared on my credit card nor have I received any confirmation email (not sure if I'll get one though). Does anyone know if they're severely backlogged for orders? -- Jason T. Nelson [EMAIL PROTECTED] http://www.jtn.cx/~jtn/ BOFH Extraordiaire Sysadmin Ombudsman GPG key 0xFF676C9E GPG key fingerprint = 6272 5482 EDDD D0A3 FED2 262A FABB 599D FF67 6C9E disclaimer: My opinions are my own. Don't bother my employer about them. Visit Winfield Public Schools at http://usd465.com - This mail sent through IMP: http://horde.org/imp/ ___ 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] softphone configs?
I noticed you had collerid not callerid in the conf file. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: 18 February 2004 19:57 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] softphone configs? k...Here you go. (that is if attachments are allowed). If not I'll find out in a minute and just send the text of the config. Thanks Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rana Dutt Sent: Wednesday, February 18, 2004 1:55 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] softphone configs? What does your sip.conf look like? Please include it in your next message in its entirety. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Wednesday, February 18, 2004 1:08 PM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] softphone configs? I've tried using the x-lite softphone as well as sjphone. I've gone over my configurations a dozen times...and I always seem to get the following error: Feb 18 11:30:16 NOTICE[1125329600]: chan_sip.c:5577 handle_request: Registration from 'Mark sip:[EMAIL PROTECTED]' failed for '192.168.5.64' FYI...I'm trying to do all my voip internally, nothing to the outside world yet. If anyone could give me an idea I'd appreciate it. Thanks. Mark ___ 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] HELP!!!! Having problems Starting Asterisk
I had this problem with an old 16bit Sound Blaster Card. Threw the card away and put in a cheap ?3.50 PCI card. Works a dream now. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Boardman Sent: 15 February 2004 23:20 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] HELP Having problems Starting Asterisk Tim Sailer wrote: On Fri, Feb 13, 2004 at 10:37:01PM +, Robert Boardman wrote: I have been trying to start asterisk all night after a reboot I keep getting this error scrolling up the screen ouch: error while writing audio data broken pipe when I go to another console there are 4 instances of mpg123 running and when I do TOP they are taking 100% CPU between them I have re installed mgp123 but it still doesn't help any Ideas? Try shutting down all * processes (including mpg123). Now, see if your audio works normally. If not, rmmod the zaptel/fx? modules, and see if that works. If not, you should start by getting your audio on the consloe to work normally first, then, check with the zap/etc modules loaded, then try * . One step at a time. Tim Thanks for the advice but I don't have any console audio device, I'm still working on it so any other advise would be appreciated, do you think I need to rebuild the system? Thanks Robb ___ 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] Jump to extension from voice menu
If you add
include = context-of-normal-extensions
at the beginning of you MENU section then this should work.
[mainmenu]
;
;main menu context with submenu
;
exten = s,1,Answer
include = default
;exten = s,2,SayDigits(${CALLERID})
exten = s,3,Background(hello_and_thank_you)
exten = s,4,Wait,t,2
exten = s,5,Goto(options,s,1)
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of bam
Sent: 11 February 2004 09:35
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Jump to extension from voice menu
Is there a way to allow a caller to enter an extension number that is more
than one digit long in a voice menu?
I want to have a menu that allows something like If you know the extension
number of the person please enter it otherwise 1 for sales, 2 for...etc
many thanks in advance,
Brian.
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RE: [Asterisk-Users] incoming call to internal user
Matteo, try: - [incoming] include = default ;default location for internal phones exten = s,1,Answer exten = s,2,Wait 10 exten = s,3,Dial(SIP/100) exten = s,4,Hangup Make sure that the context of incoming is defined in zapata.conf for pstn calls. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matteo Rancilio Sent: 09 February 2004 16:14 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] incoming call to internal user Hi Is it possible to have an incoming call forwarded directly to an internal user (we have ISDN and chan_capi)? I have internal numbers like 101,102,103,104 and so on. I need that an external user, that want to talk directly with one of us, can digit our company number and when * answer the phone waits for 5 second the input for the extension. If the extension matches the internal like 101 the call will be forwarded directly to the 101 user. If * after 5 seconds doesn't get anything the call will be routed to the operator. Is it possible without voice menu? ___ 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] New Firmware for Grandstream Phones - Supports CFG by MAC address
Have a look at http://www.plugndial.com/aps_sample.html Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of WipeOut Sent: 09 February 2004 17:03 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] New Firmware for Grandstream Phones - Supports CFG by MAC address Matthew B Marlowe wrote: The newest firmware from grandstream supports configuration by mac address. Simply upload a file cfgmac address.txt Does anyone know the format of a cfg.txt? ???R?f??)?+-?^?+$?K l? ???r???b???v(?oo?j)fj??b???j?^?+$?? ???P ? (??]j+???il? ???r?+-?w??-z Everyone has been after the format for ages, but so far I don't think anyone has it.. later.. ___ 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] The Smallest Asterisk Server Ever?
Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for that. The Linux bit is all free, and only a couple of PCB work to disenable the protection. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: 03 February 2004 18:01 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I read a report of Asterisk running on a Microsoft X-Box. That's kind of a stunt as you could buy a decent PC for the price of a Linux-capable XBox. Id's still like to see Asterisk run on very low-end hardware The Snom IP phone runs Linux inside? I assume as Linux is GPL'd Snom will supply the source code? It would be fun to install an Asterisk server in a phone. --- Panny Malialis [EMAIL PROTECTED] wrote: Does anyone have it running on a Cyclades T100 ? same as used for ntop/nbox. I was thinking of using that as an IAX-sip translator for offices with NAT. CPU MPC855T (PowerPC Dual-CPU) Memory 32MB RAM / 4MB Flash (TS100) Interfaces1 Ethernet 10/100BT on RJ45 1 RS232 Console on RJ45 RS232 Serial Ports on RJ45 Looks like fun! Although a little lacking on memory. Any comments? Panny ___ 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 = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ 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] Auto dial in Off Hook situation.
Hi all, I have looked through the wiki for any information on how to make an extension autodial another extension when it goes off hook. Anyone done this or know how it's done. regards Dave ___ 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] Auto dial in Off Hook situation.
Thanks John, I think it is not that simple. I am not using a phone but a Cisco ATA. The scenario: - User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100 (FXO))--Cisco ATA--Asterisk--Any extension The Multitech MVP100 used to connect to my old analogue switch which was set to auto call one extension. The old switch died, (rest it's soul), and I have built the * to replace, (nay superseded) it. Lot more functions for less of the greenbacks. So it is really the Cisco ATA that I need to auto call an extension. Just to cap it all I can't seem to get into the web interface of the Cisco at present, Keep getting Invalid Access. regards Dave SipPhone: - 1-747-386-2964 IaxTel: - 1-700-818-8820 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Todd Sent: 30 January 2004 14:23 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Auto dial in Off Hook situation. Hi all, I have looked through the wiki for any information on how to make an extension autodial another extension when it goes off hook. Anyone done this or know how it's done. regards Dave Depends on the phone. If you have an FXS interface, look for immediate= in your zapata.conf file. If you have an IP phone, search the vendor's documentation for PLAR (Private Line Auto Ringdown) JT ___ 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] Auto dial in Off Hook situation.
James I would have to change several other units over from proprietary to h323 that are already in the loop. I added mine to the loop so they could call for support. I have started to play with h323 on the * but not got my head round it yet. Regards Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of James Sharp Sent: 30 January 2004 18:55 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Auto dial in Off Hook situation. Thanks John, I think it is not that simple. I am not using a phone but a Cisco ATA. The scenario: - User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100 (FXO))--Cisco ATA--Asterisk--Any extension Any reason you can't use the H.323 load for the MVP200? I've not tried it in a year or so, but it mostly worked last time I tried it. ___ 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] Auto dial in Off Hook situation.
Thanks John, Found it. The Multitech's are part of a legacy system used by a new customer of mine. I just latched onto it for ease of communications, it's been in for some years now. Regards Dave ___ 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] specific to X100P with UK telephone lines
Deepak, I am using X100P on a telewest service with no problems at all. Contact me off list and I can send you a copy of my configs. [EMAIL PROTECTED] Regards Dave -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Deepakumar JVSent: 29 January 2004 06:34To: [EMAIL PROTECTED]Subject: [Asterisk-Users] specific to X100P with UK telephone lines Hello all, I got this wierd problem with X100P. When i try to dial any no over the PSTN line, i get only the dial tone. Is there any specific settings that i need to do to use X100P card with UK telephone lines? Telewest is my service provider. Is anyone using X100P in UK with telewest without any problem? could you share your settings or give me some direction? I approached digium on this and got a RMA X100P card also, still the same problem. Tried in a different system also , same problem. Wondering what would be the cause?? Regards Deepak
[Asterisk-Users] ZAP Problems
Hi all, Since my upgrade to CVS dated 14-01-2004 I am unable to call or receive calls through my ZAP channel. When calling out I get the following message: - WARNING [155667]:app_dial.c:527 dial_exec: Unable to create channel of type ZAP In zaptel.conf fxsks=1 loadzone=uk defaultzone=uk In zapata.conf language=en contect=default switchtype-euroisdn signaling=fxs_ks rxwink=300 I have done: - modprobe zaptel modprobe wcfxo ztcfg -vv results: - Zaptel Configuration Channel Map: Channel 01: FXS Kewlstart (Default) (Salves:01) 1 Channels configured Any help to resolve would be appreciated. Regards Dave ___ 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] Conf files
Hi All,
In my extensions.conf I have : -
exten = _6XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _6XXX,2,Playback(remote_unavail)
exten = _6XXX,3,Hangup
;
exten = _7XXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _7XXX,2,Playback(remote_unavail)
exten = _7XXX,3,Hangup
;
exten = _81XX,1,Playback(transfer,skip)
exten = _81XX,2,Dial(IAX2/[EMAIL PROTECTED],50,t)
exten = _81XX,3,voicemail2(u${EXTEN})
exten = _81XX,5,Goto(s|6)
exten = _81XX,103,voicemail2(b${EXTEN})
exten = _81XX,104,Goto(5)
;
exten = _80XX,1,Playback(transfer,skip)
exten = _80XX,2,Dial(SIP/${EXTEN},20,t)
exten = _80XX,3,voicemail2(u${EXTEN})
exten = _80XX,5,Goto(s,6)
exten = _80XX,103,voicemail2(b${EXTEN})
exten = _80XX,104,Goto(5)
Which is great for setting up banks of extensions.
Question Is there any way to set a range of SIP, IAX and VOICEMAIL
extensions up in the coresponding .conf files instead of: -
SIP.CONF
[8000] ; SIP Phone
type=friend
insecure=yes
host=dynamic
reinvite=no
canreinvite=no
nat=1
mailbox=8002
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=speex
allow=lpc10
;
[8001] ; SIP Phone
type=friend
insecure=yes
host=dynamic
reinvite=no
canreinvite=no
nat=1
mailbox=8003
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=ilbc
allow=speex
allow=lpc10
IAX.CONF
[8100]
type=friend
host=dynamic
secret=
disallow=all
allow=gsm
context=default
;
[8101]
type=friend
host=dynamic
secret=
disallow=all
allow=gsm
context=default
;
[8102]
type=friend
host=dynamic
secret=
disallow=all
allow=gsm
context=default
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RE: [Asterisk-Users] configuration to Grandstream via tftp
Hans, Attached is the config file I send to my Grandstream. Change IP address Phone ID to suite. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 19 January 2004 08:43 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] configuration to Grandstream via tftp Hi, Anyone know how to set up tftp server for grandstream. I gues it should be somethink like tftpserver-dir mac-address firmware.bin config.txt Is this correct ? And how should the config-file look like. ? I had search sipphone.com but did'nt find anything. /HHA _ Rethink your business approach for the new year with the helpful tips here. http://special.msn.com/bcentral/prep04.armx ___ 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 # SIP Configuration File, Plug'n'Dial APS v1.1 mac=000b820c2371 sipserver=proxy01.sipphone.com sipserver_port=5060 outboundproxy=null outboundproxy_port=null userid=8003 authenticateid=8003 codec1=PCMU codec2=PCMA codec3=G723 codec4=G729 codec5=null codec6=null silence_supporession=no voice_frames_per_tx=2 ipqos=48 vlantag=0 registration_expiration=10 local_sip_port=5060 local_rtp_port=5004 use_random_rtp_port=no stun=stun01.sipphone.com stun_port=3478 tftp_server=192.168.x.x tftp_server_port=69 send_dtmf=in-audio dtmf_payload_type=101 ntp_server=ntp01.sipphone.com time_zone=GMT-0
RE: [Asterisk-Users] configuration to Grandstream via tftp
This is the URL I got the config file from, http://www.plugndial.com/ it's on a link from the SipPhone URL. I just modified the text for my phone. There is a bit more info on there, and there is a MAC address on the top line of the file. Still just playing with this myself so don't know all the answers yet. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Hans-Henrik Andresen Sent: 19 January 2004 09:52 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] configuration to Grandstream via tftp Thanks. How is the directory structure ? or do you add all you phone to the one file cfg.txt and have it in the root of your tftp-dir ? /HHA Attached is the config file I send to my Grandstream. Change IP address Phone ID to suite. _ Find high-speed net deals comparison-shop your local providers here. https://broadband.msn.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
