[asterisk-users] DUNDi and Lua dialplan
Hello,
I would like to known how to use DUNDi with a Lua dialplan ?
In extensions.conf, we should do like these:
|[lookupdundi]
switch = DUNDi/priv
[internal]
include = dundiextens
include = lookupdundi
exten = _,2,NoOp(calling ${EXTEN})
exten = _,n,Dial(SIP/${EXTEN})
exten = _,n,Hangup()|
priority 1 is either defined in dundiextens (local registered devices)
or lookupdundi (remote)
But as in Lua there is no priority, we can't to this.
I found the following method working:
|extensions = {
internal = {
[_] = function(c,e)
app.noop('lua:: dialing exten ' .. e)
-- Goto is not working, I need to use a Local
channel
app.dial('Local/'..e..'@lookupdundi')
app.dial('SIP/'..e)
app.hangup()
end;
};
}|
But is this correct/the best one ?
Regards,
Guillaume
--
Guillaume Bourgb...@proformatique.com - proformatique
10 bis, rue Lucien VOILIN - 92800 Puteaux
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[asterisk-users] asterisk 1.8 SIP register uri: peer field ?
Hello, Looking the asterisk 1.8 API documentation (http://www.asterisk.org/astdocs/api/index.html), I see a lot of new fields for sip register uris: register = [peer?][transport://]us...@domain][:secret[:authuse...@host[:port][/extension][~expiry] But the *peer* is not explained anywhere. What it is for ? Regards, Guillaume Bour. -- Guillaume Bourgb...@proformatique.com - proformatique 10 bis, rue Lucien VOILIN - 92800 Puteaux -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChannelStateDesc: Ring ?
Martin a écrit : Ring is the state when the device sent 100 Trying after INVITE When it actually sends 180 Ringing or gets the progress or so message from another channel (when used with Dial) then the status changes to Ringing Humm. OK. So basically, it's Intended to ring... Thanks for the info. All the best, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AMI 1.0 - 1.1 with originate.
Miguel Molina a écrit : Guillaume Yziquel escribió: So what is this permission issue? Where are the changes from 1.0 to 1.1 documented? When I was testing asterisk 1.6.0.X with the AMI Originate action, I fell into the same issue as you. I found that it was that the permissions now are more fine-grained, and to have the ability to originate a call you need to set additional write permissions compared to the 1.4.X AMI. When I put the originate permission on the write settings of my AMI user, everything went fine. To find more documentarion about the changes from AMI 1.0 to 1.1 take a look of these files on your asterisk source code: UPGRADE-1.6.txt doc/manager_1_1.txt Hope it solves your issue. It pretty well did. Thanks a lot. -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI 1.0 - 1.1 with originate.
Hello. I wrote a piece of code for AMI 1.0, to originate some calls. The code is (in OCaml): string_of_lines [ Action: originate; (Channel: ^(Configuration.dial_campaign_item campaign_item)); WaitTime: 30; CallerId: appel ; Exten: receiver; Context: receiving; Priority: 1; Async: true; (ActionID: ^unique_identifier)] It used to work fine with the AMI 1.0, but now, with 1.1., I get Response: Error ActionID: 1256035727.29741096 Message: Permission denied So what is this permission issue? Where are the changes from 1.0 to 1.1 documented? All the best, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChannelStateDesc: Ring ?
Hello. I've experience a rather surprising behaviour of the AMI 1.1 Event: Newstate^M Privilege: call,all^M Channel: SIP/XX-089c63b8^M ChannelState: 4^M ChannelStateDesc: Ring^M CallerIDNum: ^M CallerIDName: Y^M Uniqueid: 1256089773.59^M Usually ChannelStateDesc gives me 'Ringing' but sometimes it only gives me 'Ring', which drives my application to 100% CPU (OK, not a very resilient app for now...) My question is: is 'Ring' a specific state for a channel, or is it a known bug in the AMI? All the best, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 2 IPs for an Asterisk server.
Hello. I've been setting up an Asterisk server, and I am now supposed to move it to a different network than the one it was set on. I'd like to give the server 2 IP address: -1- The first IP address is the IP it will have on the LAN, meaning that softphones will register to the Asterisk server using this 1st IP. -2- The second IP is the one that it will use to connect to the remote VoIP provider, which is using another network range than the LAN where I have my softphones. The default gateway would be the one of this second network address range. No NAT involved anywhere in this setup. Is it possible to do such a thing with Asterisk? Does it need really special tweaking of Asterisk conf files? -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 2 IPs for an Asterisk server.
Jorge Gutiérrez a écrit : Yes it is possible, the only thing that you need to do is to configure correctly your network routes, if your ip devices are on the same net of your elastix you wont need to do any route configuration. Just leave the default gateway for your wan provider, it should work without any trouble Thank you for this valuable information. -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] A in ACL of sip show peers.
Hello. ubuntu*CLI sip show peers Name/username HostDyn Nat ACL Port Status voipprovider xxx.xxx.xxx.xxx A 5060 Unmonitored I've ben trying to connect an asterisk server to a voip provider, and I'm currently wondering what the 'A' in the ACL field of the 'sip show peers' command might be. I've been unable to find this information on the net. Help would be appreciated. -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stop / Resume in Dialplan / AMI
Hello. I'd like to know if the two following functionalities are available in Asterisk. -1- A stop/wait/halt functionality in the Dialplan. Like: exten = myexten, n, Halt where execution of the dialplan would wait indefinitely. I guess a Wait would be OK, but I'd like this wait to wait indefinitely. -2- A Goto functionality from the AMI: You give the channel, and you can ask it to change its priority. -3- Or a WaitForEvent: The AMI sends an event, and the Dialplan resumes dialplan execution after this event. These are a few questions I have about AMI / Dialplan asynchronous integration, and guidance would be appreciated. All the best, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incremented UniqueId
Hi. I've been using the Asterisk Manager Interface to originate calls from Console/dsp. I get the following form the server. Response: Success Message: Originate successfully queued Uniqueid: asterisk-3301-1252055630.26701 Event: Newchannel Privilege: call,all Channel: Console/dsp State: Down CallerIDNum: unknown CallerIDName: unknown Uniqueid: asterisk-1252055630.26702 I'm really wondering how the Uniqueid works. Why is it incremented? What is the dot for in the Uniqueid? Is there any thorough documentation on what the statements of the AMI mean? All the best, -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incremented UniqueId
Steve Howes a écrit : On 4 Sep 2009, at 10:36, Guillaume Yziquel wrote: Uniqueid: asterisk-1252055630.26702 I'm really wondering how the Uniqueid works. Why is it incremented? What is the dot for in the Uniqueid? The bit before the dot is a unix timestamp (Fri, 04 Sep 2009 09:13:50 GMT in this case). The bit after the dot is.. randomish.. but sequential. S Randomish, but sequential... So how can I programmatically link my Originate successfully message to my Newchannel event? -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Originate calls with AMI.
Hello. I've been trying to use the AMI to originate phone calls. I'm trying to call the SIP phone 'zoiper' with the SIP phone 'yziquel'. So, the AMI interaction is: Action: originate Channel: SIP/zoiper Exten: yziquel Priority: 1 Timeout: 30 Context: internal Response: Error Message: Originate failed Event: Newchannel Privilege: call,all Channel: SIP/zoiper-019a3000 State: Down CallerIDNum: unknown CallerIDName: unknown Uniqueid: asterisk-1251987055.7 Event: Newcallerid Privilege: call,all Channel: SIP/zoiper-019a3000 CallerID: Unknown CallerIDName: Unknown Uniqueid: asterisk-1251987055.7 CID-CallingPres: 0 (Presentation Allowed, Not Screened) Event: Hangup Privilege: call,all Channel: SIP/zoiper-019a3000 Uniqueid: asterisk-1251987055.7 Cause: 0 Cause-txt: Unknown And then the 'zoiper' softphone starts ringing continuously. It says Incoming Call from asterisk and not from 'yziquel'. Moreover when I pick up the phone it says You are now talking to asterisk, and then Zoiper closes the call immediately. There's surely something I do not get right here, and I'd appreciate some help. All the best, Guillaume YZiquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Originate calls with AMI.
Hello. Matt Riddell a écrit : To start with I'd do (just rearranging but makes me feel better): Action: originate Channel: SIP/zoiper Context: internal Exten: yziquel Priority: 1 Timeout: 30 Callerid: yziquel Thank you for your answer. But also, are you sure that the extension yziquel exists in the internal context? Yes, it does. I finally got it right (no rearrangement) with Action: originate Channel: SIP/zoiper WaitTime: 30 CallerId: yziquel Exten: yziquel Context: internal Priority: 1 Somehow surprised that the only needed change was to change Timeout to Waitime... type the following: dialplan show internal Here it is: seldon*CLI dialplan show internal [ Context 'internal' created by 'pbx_config' ] '500' = 1. Verbose(1|Echo test application) [pbx_config] 2. Echo() [pbx_config] 3. Hangup() [pbx_config] 'yziquel' = 1. Verbose(1|Extension yziquel) [pbx_config] 2. Dial(SIP/yziquel|30) [pbx_config] 3. Hangup() [pbx_config] 'zoiper' = 1. Verbose(1|Extension zoiper)[pbx_config] 2. Dial(SIP/zoiper|30)[pbx_config] 3. Hangup() [pbx_config] -= 3 extensions (9 priorities) in 1 context. =- Thanks a lot. -- Guillaume Yziquel http://yziquel.homelinux.org/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bottlenecks with my asterisk setup.
Hello. I've been seting up a small VoIP setup, with roughly 5 persons, doing essentially some Meetme conferences. People have been experiencing some quality problems with the sound. Essentially delay, and some tolerable echo. I'd appreciate advice on how to troubleshoot this issue. What could be the most common reasons behind this? Please feel free to ask for more relevant details. All the best, Guillaume Yziquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bottlenecks with my asterisk setup.
Guillaume Yziquel a écrit : Hello. I've been seting up a small VoIP setup, with roughly 5 persons, doing essentially some Meetme conferences. People have been experiencing some quality problems with the sound. Essentially delay, and some tolerable echo. I'd appreciate advice on how to troubleshoot this issue. What could be the most common reasons behind this? Please feel free to ask for more relevant details. Another question: should I expect these issues to be less important if I switch to a Zaptel configuration instead of only VoIP? All the best, Guillaume Yziquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP server behind NAT
Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The Asterisk server connects to a remote VoIP provider via SIP. The extensions.conf and sip.conf follow below. I contacted the provider to see if there was a specific problem, and I was advised that my asterisk was not using standard ports for the transit of voice (I assume they mean RTP on UDP). They're telling me that, normally, the port 5004 should be used, or ports above 1. However for one of my outbound calls, they see me using port 3030. Could someone advise me on the steps to follow, or to documentation on this issue? Does this sound like a NAT issue? All the best, Guillaume Yziquel. Here's the beginning of the sip.conf file: [general] context=default ; Default context for incoming calls ;allowguest=no ; Allow or reject guest calls (default is yes, this can also be set to 'osp' ;realm=mydomain.tld ; Realm for digest authentication 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) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ;domain=mydomain.tld; Set default domain for this host ;domain=mydomain.tld,mydomain-incoming ;domain=1.2.3.4 ; Add IP address as local domain ;allowexternalinvites=no; Disable INVITE and REFER to non-local domains ;autodomain=yes ; Turn this on to have Asterisk add local host ;pedantic=yes ; Enable slow, pedantic checking for Pingtel ;tos=184; Set IP QoS to either a keyword or numeric val tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpiry=3600 ; Max length of incoming registration we allow defaultexpiry=120 ; Default length of incoming/outgoing registration ;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY ;checkmwi=10; Default time between mailbox checks for peers ;vmexten=voicemail ; dialplan extension to reach mailbox sets the ;videosupport=yes ; Turn on support for SIP video ;recordhistory=yes ; Record SIP history by default disallow=all; First disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=gsm ; musicclass=default ; Sets the default music on hold class for all SIP calls language=en ; Default language setting for all users/peers relaxdtmf=yes ; Relax dtmf handling rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity trustrpid = no ; If Remote-Party-ID should be trusted sendrpid = yes ; If Remote-Party-ID should be sent progressinband=no ; If we should generate in-band ringing always useragent=My Asterisk ; Allows you to change the user agent string promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address ;usereqphone = no ; If yes, ;user=phone is added to uri that contains dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 ;compactheaders = yes ; send compact sip headers. ;sipdebug = yes ; Turn on SIP debugging by default, from ;subscribecontext = default ; Set a specific context for SUBSCRIBE requests ;notifyringing = yes; Notify subscriptions on RINGING state ;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected, ;regcontext=sipregistrations ;registertimeout=20 ; retry registration calls every 20 seconds (default) ;registerattempts=10; Number of registration attempts before we give up callevents=no ; generate manager events when sip ua performs events (e.g. hold) externip=The_IP_of_my_router; Address that we're going to put in outbound SIP messages ;externhost=foo.dyndns.net ; Alternatively you can specify an ;externrefresh=10 ; How often to refresh externhost if localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks localnet=10.0.0.0/255.0.0.0 ; Also RFC1918 localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation localnet=169.254.0.0/255.255.0.0 ;Zero conf local network nat=yes ; Global NAT
[asterisk-users] Outbound calls drop after 15 to 30 seconds.
Hello. I've set up and configured an Asterisk server to make SIP phone calls to external classic phones. However, it happens that after 15 or 30 seconds, the phone call drops. The SIP session still seems valid, but no sound comes through any more. How would you go through to troubleshoot this issue? All the best, Guillaume Yziquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound calls drop after 15 to 30 seconds.
Steve Totaro a écrit : On Mon, Aug 3, 2009 at 8:20 AM, Guillaume Yziquel guillaume.yziq...@citycable.ch wrote: Hello. I've set up and configured an Asterisk server to make SIP phone calls to external classic phones. However, it happens that after 15 or 30 seconds, the phone call drops. The SIP session still seems valid, but no sound comes through any more. How would you go through to troubleshoot this issue? All the best, Guillaume Yziquel. Make sure you have canreinvite set to no. It was already set to 'no' Also, you may need to put an answer() in before your dial, I have dealt with that strangeness, call always drop at exactly 30 seconds. Putting exten = _X.,n,Answer() in the dialplan doesn't change anything. That solution worked for me, but I could see how it could mess up CDRs and billing for some applications. Maybe I'm having a different issue than you've been experiencing. What's rather painful is that nothing appears to show in the Asterisk CLI when this happens since it's obviously not a problem with the SIP connection. How could I monitor the voice going in and out? All the best, Guillaume Yziquel. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky
-Original Message- From: Mark Ackroyd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com CC: Subject: RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky Sent: mer., 03 mai 2006 09:15:13 GMT Received: mer., 03 mai 2006 09:18:20 GMT Read: mer., 03 mai 2006 10:07:50 GMT When I hit *1 in my system, I got a beep to let me know that the recording started. Is this not happenning to you? No ! , it doesn't. Most of the time it doesn't pick up the *1. Hi! Just make sure to hit *1 very quickly... I mean with minimum delay between * and 1. Guillaume ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: RE: RE: [Asterisk-Users] Dial Option wW picking up the *1 is abitflaky
I think you can define it in features.conf featuredigittimeout = 500 ; Max time (ms) between digits for ; feature activation. Default is 500 Check http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf Hope this helps... Guillaume Thanks Guillaume. What's the maximum allowed delay? I there a way of setting it? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillaume de Lafontaine Sent: 03 May 2006 14:59 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bitflaky -Original Message- From: Mark Ackroyd [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com CC: Subject: RE: [Asterisk-Users] Dial Option wW picking up the *1 is a bit flaky Sent: mer., 03 mai 2006 09:15:13 GMT Received: mer., 03 mai 2006 09:18:20 GMT Read: mer., 03 mai 2006 10:07:50 GMT When I hit *1 in my system, I got a beep to let me know that the recording started. Is this not happenning to you? No ! , it doesn't. Most of the time it doesn't pick up the *1. Hi! Just make sure to hit *1 very quickly... I mean with minimum delay between * and 1. Guillaume ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Re: [Asterisk-Users] Recommendations on a WiFi phone for *?
Hi I just discovered an interesting product line. Not tested yet... http://www.paesys.com/fr/lecteurs_VoIP_WiVoip_VideoIP_GSM.htm In french, sorry... Any feedback ? --- Guillaume de Lafontaine ___ D W A M ___ -Original Message- From: Jean-Michel Hiver [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com CC: Subject: Re: [Asterisk-Users] Recommendations on a WiFi phone for *? Sent: mar., 10 janv. 2006 13:17:02 GMT Received: mar., 10 janv. 2006 13:18:11 GMT Read: mar., 10 janv. 2006 13:22:28 GMT Rupert Gregory a écrit : Once you've finished drooling over the UTStarcom you can start drooling over the Linksys WIP330 http://ces.engadget.com/2006/01/07/linksys-wip330-in-da-house-but-you-cant-have-one/ VERY nice phone in my opinion. I dunno... it looks like a cell phone, except it's not one. It would be nice if it was a dual GSM / wifi phones which transparently switch to VoIP when you have a strong enough signal. This way, it would provide all the cost savings of VoIP with the convenience of GSM calls. Of course it would need to display a big bright icon to let the user know when they are not on wifi / voip since GSM providers are pretty expensive... Also, voicemail would become very nice. Get out of the office, the SIP register times out, and you're on voicemail. (of course you could also forward the call to the GSM number although it might be a little more expensive). X is out of the office kind of message would actually make sense... Get back in the office, the phone registers, and you get MWI. Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PSTN CALLER ID FRANCE TELECOM
Thanks Olivier Actually, today I have tried the patch provided by [http://www.lusyn.com/asterisk/patches.html] as indicated on the Asterisk and UK Caller ID page [http://www.voip-info.org/wiki/view/Asterisk+and+UK+Caller+ID] It works now ! My card is a X100P clone, (Tiger3XX Modem/ISDN) The relevant part of zapata.conf is now : usecallerid=yes cidsignalling=v23 cidstart=usehist Guillaume ___ --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
[Asterisk-Users] PSTN CALLER ID FRANCE TELECOM
Hi all I try to get the caller id of a incoming call through a X100P generic card. I have tried many configuration on the zapata.conf, but i never succeed to have a correct CALLERIDNUM. What is the cid signaling provided by FranceTelecom (v23 ?) Is there some specific stuff to do ? Could you help me please ? Guillaume The provider is france telecom, The card is a X100P asterisk -V = 1.0.9 The error message is : Oct 10 20:17:02 WARNING[702]: chan_zap.c:5476 ss_thread: Calleerror on channel 'Zap/1-1' my zapata.conf is ~ context=from-ft language=fr signalling=fxs_ks busydetect=yes busycount=1 callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes usecallerid=yes echocancel=yes echocancelwhenbridged=yes rxgain=6.0 txgain=4.0 immediate=no callerid=asreceived musiconhold=default channel = 1 ___ --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
[Asterisk-Users] Eyebeam Video+Nat
Hi, I test the video on asterisk with eyebeam. When I use a public IP for the softphone, the video work. However, when I test eyebeam under nat the video doesnt work. I use a routeur linksys WRT54G. I try also to configure my laptop under DMZ for redirect all the traffic IP and the video doesnt work too. Can you help me please? Sincerely, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk wont accept tone from Norstar 3X8 ATA port ?
Hi all, I have a very simple setup - 2 ATA modules connected to my Norstar 3x8 extension 25 and 26 - 4 PCI FXO card in my asterisk box, two of then are connected to those extension 25 and 26. I have a context for those 2 ata modules simply to get my voicemail. From my Norstar telephone if i dial ext 25 or 26 I get into the correct context and get into voicemailmain. But, when I enter tone from my Norstar tel asterisk dont see anything. I unplug asterisk from my ATA modules and replace * with a ordinary telephone and I ear the tone that my norstar send. What could be the reason for * to not understand those tone ? Any pointer would be very appreciated ! Here more details about my setup: *CLI show version Asterisk 1.0.2 built by [EMAIL PROTECTED] on a i686 running Linux Tel. Nortel Norstar 3X8 DR 5.1 4 OEM X100P - FXO PCI Card TIA Guillaume ___ 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] ztdummy on Gentoo 2.6.10 Box
The ztdummy requires zaptel to be loaded. You don't need any zaptel devices just the kernel module. Get zaptel then follow the instructions on the wiki @ http://www.voip-info.org/wiki-Asterisk+timer+ztdummy i.e.: ztdummy and Kernel 2.6 Recent CVS versions (as of July 2004) also have zaptel modules which work with 2.6 kernels (type make linux26 to compile them). The 2.6 version of the ztdummy module is completely different to the 2.4 version. Firstly it does not rely on the USB hardware being there; instead it uses the PC hardware's clock (which under 2.6 kernels can be set to generate interrupts at the required precision), so it can be used on any machine, rather than only those with the right USB hardware. Effectively the 2.6 version of ztdummy does the same job as zaprtc does for 2.4 kernels. I have been using ztdummy for 2.6 for a while now, and have had no problems with it. To install, simply checkout zaptel from the Asterisk CVS and do the following: - cd /usr/src/zaptel * READ /usr/src/zaptel/README.udev and follow the steps * check modules on: /etc/sysconfig/zaptel if you have no digium hardware comment out all modeules except ztdummy. - make linux26 - modprobe zaptel - modprobe ztdummy Cheers, Guills -Original Message- From: Richard J. Sears [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 1:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztdummy on Gentoo 2.6.10 Box Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy returned this: WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting ztdummy (/lib/modules/2.6.10-gentoo-r6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy Here is what my dmesg says: zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table ztdummy: Unknown symbol zt_receive ztdummy: Unknown symbol zt_transmit ztdummy: Unknown symbol zt_unregister ztdummy: Unknown symbol zt_register I DO NOT have any Zaptel devices in my system and I VIed the makefile and uncommented the ztdummy as instructed. I guess my question is - what the heck is happeneing. Why is ztdummy trying to load zaptel which I do not have in my system..? Any help would be greatly appreciated. Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching. ___ 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] ztdummy on Gentoo 2.6.10 Box
This last error message now means zaptel is compiled (which it wasn't before) but now it can't find that table. You need to enable it in your kernel config it seems. As mentionned here: http://lists.digium.com/pipermail/asterisk-dev/2004-December/008303.html The option to turn this on is in the last submenu of the kernel config menu. Guills -Original Message- From: Richard J. Sears [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 4:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Chamberland-Larose, Guillaume Subject: Re: [Asterisk-Users] ztdummy on Gentoo 2.6.10 Box Hi Guills, I followed the instructions exactly as described on the voip pages as well as your instructions below. I still get the same errors. pbx01 zaptel # modprobe zaptel FATAL: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) DMESG: zaptel: Unknown symbol crc_ccitt_table And of course ztdummy fails to load becuase the zaptel does not load. Thanks On Mon, 14 Feb 2005 13:55:04 -0800 Chamberland-Larose, Guillaume [EMAIL PROTECTED] wrote: The ztdummy requires zaptel to be loaded. You don't need any zaptel devices just the kernel module. Get zaptel then follow the instructions on the wiki @ http://www.voip-info.org/wiki-Asterisk+timer+ztdummy i.e.: ztdummy and Kernel 2.6 Recent CVS versions (as of July 2004) also have zaptel modules which work with 2.6 kernels (type make linux26 to compile them). The 2.6 version of the ztdummy module is completely different to the 2.4 version. Firstly it does not rely on the USB hardware being there; instead it uses the PC hardware's clock (which under 2.6 kernels can be set to generate interrupts at the required precision), so it can be used on any machine, rather than only those with the right USB hardware. Effectively the 2.6 version of ztdummy does the same job as zaprtc does for 2.4 kernels. I have been using ztdummy for 2.6 for a while now, and have had no problems with it. To install, simply checkout zaptel from the Asterisk CVS and do the following: - cd /usr/src/zaptel * READ /usr/src/zaptel/README.udev and follow the steps * check modules on: /etc/sysconfig/zaptel if you have no digium hardware comment out all modeules except ztdummy. - make linux26 - modprobe zaptel - modprobe ztdummy Cheers, Guills -Original Message- From: Richard J. Sears [mailto:[EMAIL PROTECTED] Sent: Monday, February 14, 2005 1:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztdummy on Gentoo 2.6.10 Box Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy returned this: WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) WARNING: Error inserting zaptel (/lib/modules/2.6.10-gentoo-r6/misc/zaptel.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error inserting ztdummy (/lib/modules/2.6.10-gentoo-r6/misc/ztdummy.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for ztdummy Here is what my dmesg says: zaptel: Unknown symbol crc_ccitt_table zaptel: Unknown symbol crc_ccitt_table ztdummy: Unknown symbol zt_receive ztdummy: Unknown symbol zt_transmit ztdummy: Unknown symbol zt_unregister ztdummy: Unknown symbol zt_register I DO NOT have any Zaptel devices in my system and I VIed the makefile and uncommented the ztdummy as instructed. I guess my question is - what the heck is happeneing. Why is ztdummy trying to load zaptel which I do not have in my system..? Any help would be greatly appreciated. Thanks ** Richard J. Sears Vice President American Internet Services [EMAIL PROTECTED] http://www.adnc.com 858.576.4272 - Phone 858.427.2401 - Fax INOC-DBA - 6130 I fly because it releases my mind from the tyranny of petty things . . Work like you don't need the money, love like you've never been hurt and dance like you do when nobody's watching
RE: [Asterisk-Users] Proper Contexts in extensions.conf
What about... [incoming] Include = internal [sip-extensions] Include = internal Include = long-distance [internal] ... internal extensions ... [long-distance] ... -Original Message- From: Max Clark [mailto:[EMAIL PROTECTED] Sent: Thursday, February 10, 2005 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Proper Contexts in extensions.conf Hi all, I am looking for examples of the extensions.conf that puts all incoming calls into a context where extensions can be dials, and all phones in a context where extensions and outside calls can be dialed. i.e. I have seen: [incoming] include = sip-extensions [sip-extensions] include = longdistance [longdistance] Doesn't this allow any internal callers to make external calls? How do you properly set this up? Thanks, Max -- Max Clark max [at] clarksys.com http://www.clarksys.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] SIP ActiveX
Why do people want to develop more softphones? There's already lots of softphone projects out there that could use a hand, if you want to work on one. Starting from scratch will just add one more half baked softphone to the growing list of unuseful Open Source applications. Especially since you consider using an ActiveX control :) Guills -Original Message- From: JOAO CARLOS MOURA [mailto:[EMAIL PROTECTED] Sent: Wednesday, February 09, 2005 7:47 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] SIP ActiveX I search a ActiveX to develop one softphone SIP with codec G723. Who can help me? Thank´s João Carlos Moura ___ 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] Limit MOH processes
Is it possible your MoH class isn't set properly for that call? It should say class 'default' not class ''. But then again, IANAME (I Am Not A Music-on-hold Expert) ;) Guills -Original Message- From: Stefan Gofferje [mailto:[EMAIL PROTECTED] Sent: Saturday, February 05, 2005 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Limit MOH processes Chamberland-Larose, Guillaume schrieb: - use one of the many patches for native MOH without mpg123 Well, doesn't that mean, I have to convert all the mp3s to another bitrate/format? *sigh* I'm using a whole bunch of my mp3s and I didn't have to convert any of them. It seems to be working fine. All I had to do is install the addon. I have got latest asterisk-addons w/ format_mp3. Installed and configured it exactly as said in the WiKi. ; ; Music on hold class definitions ; [classes] [moh_files] default = /var/lib/asterisk/moh-native Problem: Feb 6 00:35:26 WARNING[8815]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class '') on channel SIP/6004-7dae Do you have a hint on that? Can I have multiple classes like with mpg123? 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] Asterisk causing server to hang ... any hints?
If the asterisk process is hung up you should be able to debug it. If
the whole machine is hung up this is a totally different issue isn't it.
If you're running linux and the machine locks up that often, you must
have a hardware problem. Or maybe you just think the machine is locked
up while it actually isn't.
You might want to look on google and the wiki about deadlocks, I've seen
a lot of information around on how to debug them and report them.
Guills
-Original Message-
From: beonice [mailto:[EMAIL PROTECTED]
Sent: Tuesday, February 08, 2005 3:41 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk causing server to hang ...
any hints?
I am trying to set up a simple Asterisk server. All it's
going to do for now is to act as my voicemail box. I've got a
DID from Voicepulse, and am using IAX (I'll get to SIP
someday when I want to circumvent the phone company for
long-distance, but for now I'd be happy to get a trial
version of Asterisk running).
So far, I've managed to set up voicemail.conf,
extensions.conf and iax.conf so that incoming calls are
answered and a voicemail can be left. Initially I did not
have a handler for the timeout case, and Asterisk would
complain about the missing handler and occasionally would
crash the server after several messages had been left (I'm
the only one testing it, and the server has nothing else
running, just Asterisk), requiring a hard reboot.
To avoid this, I put in a handler for the timeout.
Now, Asterisk crashes the server as soon as I leave myself a
message! This is irritating. Hard reboots everytime someone
leaves me a voicemail is not going to be something I can
trust if I go on vacation (hopefully someday soon).
Any ideas what I am doing wrong?
Here's the change I made to my extension file (this used to
crash occasionally):
[old-context]
exten = ,2,VoiceMail,u
exten = ,3,Hangup
exten = ,102,VoiceMail,b
exten = ,3,Hangup
exten = ,103,Hangup
[new-context] ;; crashes all the time!
exten = ,2,VoiceMail,u
exten = ,3,Hangup
exten = ,102,VoiceMail,b
exten = ,3,Hangup
exten = ,103,Hangup
exten = ,1,Playback(transfer,skip)
exten = ,2,Ringing
exten = ,3,Wait(2)
exten = ,4,VoiceMail,u
exten = ,104,VoiceMail,b
exten = 1001,1,Ringing
exten = 1001,2,Wait(2)
exten = 1001,3,VoicemailMain
exten = _NXXNXX,1,Background(beep)
;exten = _NXXNXX,2,SayDigits(${EXTEN}) ;exten =
_NXXNXX,3,Goto(testdtmf|s|1) exten = t,1,Ringing exten
= t,2,Hangup exten = i,1,Ringing exten = i,2,Hangup exten
= a,1,VoicemailMain,EXTEN exten = a,2,Hangup
I thought that using the
exten = t,2,Hangup and the
exten = i,2,Hangup
would cause Asterisk to hang up on timeout, but obviously I
misunderstood. By the way, what _does_ the 'a' handler do,
anyway? I believe 'i' is for invalid key and 't' is for
timeout. I'm not sure what 'a'
does.
The last time I tried to leave myself voicemail, here's the
output I got. And yes, it crashed. I've got Asterisk running
at a level of 4 in terms of verbosity, i.e., I start it with
/usr/sbin/asterisk -cp
-- Playing 'beep' (language 'en')
Feb 8 14:07:29 DEBUG[4195]: chan_iax2.c:5310
socket_read: Ooh, voice format changed to 4
== CDR updated on
IAX2/[EMAIL PROTECTED]:4569/1
-- Executing
Playback(IAX2/[EMAIL PROTECTED]:4569/1,
transfer|skip) in new stack
-- Playing 'transfer' (language 'en')
-- Executing
VoiceMail(IAX2/[EMAIL PROTECTED]:4569/1,
u) in new stack
Feb 8 14:07:42 DEBUG[4195]: app_voicemail.c:1381
leave_voicemail:
voicemail/voicepulse_connect_context//unavail
doesn't exist, doing what we can
-- Playing 'vm-theperson' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'digits/7' (language 'en')
-- Playing 'vm-isunavail' (language 'en')
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
Feb 8 14:07:53 DEBUG[4195]: app.c:549
ast_play_and_record: play_and_record: None,
/var/spool/asterisk/voicemail/voicepulse_connect_context//
INBOX/msg0001,
'wav49|gsm|wav'
Feb 8 14:07:53 DEBUG[4195]: app.c:566
ast_play_and_record: Recording Formats: sfmts=wav49
-- x=0, open writing:
/var/spool/asterisk/voicemail/voicepulse_connect_context//
INBOX/msg0001
format: wav49, 0x814ded0
-- x=1, open writing:
/var/spool/asterisk/voicemail/voicepulse_connect_context//
INBOX/msg0001
format: gsm, 0x814dff0
-- x=2, open writing:
/var/spool/asterisk/voicemail/voicepulse_connect_context//
INBOX/msg0001
format: wav, 0x814e100
-- User ended message by pressing #
-- Playing 'auth-thankyou' (language 'en') localhost*CLI
At this point, the server itself seems to hang. I can do
RE: [Asterisk-Users] Limit MOH processes
You could try to use the native mp3 support for MOH if you really want mp3 support. It is a lot better than using mpg123 IMHO. mpg123 kept doing nasty things to my system :) See http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20musicon hold.conf there is a section about the native support. Guillaume -Original Message- From: Stefan Gofferje [mailto:[EMAIL PROTECTED] Sent: Friday, February 04, 2005 1:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Limit MOH processes Hi folks, is there a way to limit the number of spawned mpg123 processes. I have 22 mpg123 in my tasklist, consuming 97% of CPU power. That seems a little too much for me, considering, I have just 4 external lines... 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] Limit MOH processes
- use one of the many patches for native MOH without mpg123 Well, doesn't that mean, I have to convert all the mp3s to another bitrate/format? *sigh* I'm using a whole bunch of my mp3s and I didn't have to convert any of them. It seems to be working fine. All I had to do is install the addon. Guills ___ 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] Calling Asterisk Autoattendant With SIP Phone
I believe the web page should be modified to include a
huge, red, bold, blinking "please read the asterisk handbook available here and
search the wiki and mail archives before you post a message to the list". That
would prevent so many questions on how and where to start when first installing
asterisk. :s
So, I would suggest you check out the asterisk handbook
here: http://www.digium.com/handbook-draft.pdf
Page 56 to 61 explain in lots of detail and give a working
example of sip.conf with 1 phone and 1 voip provider. The whole thing is good to
read though so you might as well read the whole thing (quickly)
hehe.
The handbook assumes you know nothing about asterisk and
pretty much everything else. You shouldn't have to spend more than 15 minutes
configuring this.
Guills
From: Matt Waterman [mailto:[EMAIL PROTECTED]
Sent: Wednesday, February 02, 2005 7:08 PMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Calling
Asterisk Autoattendant With SIP Phone
I'mtrying to get into the world of Asterisk
in order to use the voicemail and autoattendat features (and more stuff later)
with a Redcom switch. But, I've only started and haven't gotten to that yet.
At this point my solitary goal is to talk to the autoattendant via an SIP
phone (SJPhone). I've spent countless hours trying to find the documentation I
need to accomplish my goals but everything I find always assumes so much and
I'm left lost. Plus I haven't found a thing about setting up Asterisk as an
SIP server.
I installed the [EMAIL PROTECTED] package, so I can edit all the
config files through HTTP and I can use AMP.
I've tried 'dialing' tothe IP address of
the Asterisk machine with SJPhone but the call is rejected ("number not
available"). Now, how do I specify an extension number when I
'dial'?
Thanks for any help :/
Matt
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[Asterisk-Users] Speex pass through on SIP
Title: Speex pass through on SIP Hi, I've seen some answers to this on the mailing list archives but nothing that seems like the right answer. What I want is for 2 SIP phones to use speex to talk to each other through 2 asterisk boxes (linked over IAX2) while only supporting ulaw on the asterisk boxes themselves. I think a diagram will help ;) SIP1 -- *1 -- IAX2 link -- *2 -- SIP2 I want any calls from SIP1 to SIP2 or SIP2 to SIP1 to be able to use speex OR ulaw (depending on network status) I want any calls to *1 or *2 to use ulaw only (VM and other features) since those should be over LAN anyway. I want the IAX2 link (which is over the internet) to transmit whatever the SIP phones use (i.e. not going from speex - ulaw then back ulaw-speex on the other side) I need to make sure that if SIP1 puts SIP2 on hold, *1 won't try to send MOH using speex. In fact, if * is out of the loop it shouldn't be able to put calls on hold right? Any idea what the setup I want is for both sip phones and both sides of the IAX2 connection? Guills ___ 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] FWD and IAX2
Hi,
Just so others find it if they google. I got it to work, using the same
config I had before (similar to the one below), by going to the FWD web
page and removing my IAX2 account and creating it again. (By unchecking
the use IAX2 box, saving, checking it again and saving again)
Also, it seems it doesn't like me if I don't use trunk mode, or maybe I
just wasn't lucky when I tried. Everything works fine with trunk mode
now.
Thanks guys,
Guills
From: Gonzalo Gasca [mailto:[EMAIL PROTECTED]
Sent: Friday, January 28, 2005 10:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FWD and IAX2
Guillaume
Check this good link:
http://www.freeworlddialup.com/advanced/iax
Here is my config that works:
iax.conf
register = 421058:[EMAIL PROTECTED] ;FWD Number
421058
[iaxfwd] ; inbound connections from FWD ;it has to be
'iaxfwd' or it won't work
type=user
auth=rsa
inkeys=freeworlddialup
disallow=all
allow=ulaw ; FWD only support ulaw
context=fwd-incoming
[fwd-gw] ; outbound connections to FWD
type=peer
auth=md5
secret=pantera
username=421058
qualify=yes
host=iax2.fwdnet.net
disallow=all
allow=ulaw
callerid=Gonzalo Gasca421058
extensions.conf
;Free World Dialup
FWDUSERID=421058
FWDUSERNAME=Gonzalo Gasca
FWDGW=IAX2/[EMAIL PROTECTED]
;***
;To Free World Dialup
;***
[fwd-users]
exten = _7.,1,SetCIDNum(${FWDUSERID}) ; To
dial FWD I enter 7 first...
exten = _7.,2,SetCIDName(${FWDUSERNAME})
exten = _7.,3,Dial(${FWDGW}/${EXTEN:1},60,r)
exten = _7.,4,Hangup
;***
;From FWD
;***
[fwd-incoming]
;Incoming calls from FWD to ring SIP Extension 100
exten = _421058,1,Dial(SIP/100,20)
exten = _421058,2,Voicemail(u100)
exten = _421058,102,Voicemail(b100)
exten = _421058,103,Hangup
bye
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RE: [Asterisk-Users] IAX registration keep alives
I think qualify=1000 should work. Take a look at: http://www.voip-info.org/wiki-Asterisk+config+iax.conf Guills From: Liaan vd Merwe [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 01, 2005 4:09 AMTo: Asterisk-Users@lists.digium.comSubject: [Asterisk-Users] IAX registration keep alives hallo all could anyone tell me how to get the * to send keepalive packets over a registration "trunk" or how to increase the amount I'm having natting issues, (the machine is siting behind 2 nat firewalls) thanks liaan Do you Yahoo!?Yahoo! Search presents - Jib Jab's 'Second Term' ___ 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] Crash: Call from IAX-client to a distribution where the IAX-Client is in
Hmm. By the way, please don't post bugs to asterisk-dev as I've been told : That list if for on-going development. That sounds like a bug I encountered in 1.0.5. There is a division by zero bug in chan_iax2.c introduced somewhere after 1.0.4 I believe and currently fixed in HEAD. (They've given me enough shit for posting the bug while it was fixed in HEAD already. No need to mention it again.) Run asterisk in gdb and see if it is actually the same bug. If it is, get cvs HEAD and you should be fine. You should see something like this in gdb if it is: [Switching to Thread 245775 (LWP 23251)] 0x41154918 in calc_timestamp (p=0x816b710, ts=0, f=0x424eef24) at chan_iax2.c:2896 Guills -Original Message- From: Stefan Gofferje [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 01, 2005 3:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-dev@lists.digium.com Subject: [Asterisk-Users] Crash: Call from IAX-client to a distribution where the IAX-Client is in Hi folks, I encountered a reproduceable crash case: [extensions.conf] exten = 6000,1,Dial(SIP/6000,60,rt) exten = 6001,1,Dial(SIP/6001,60,rt) exten = 6002,1,Dial(IAX2/[EMAIL PROTECTED],60,rt) exten = 8004,1,Dial([EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/6002@ internal,60,rt] 6002 is an IAX Softphone (tested firefly, IAX-Phone, IAXComm) When 6002 dials 8004, asterisk quits without further notice and log-entry. When 8004 is dialled from any other source, everything is fine. Any clues? 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 registration keep alives
I don't know how you're setting it up, but usually a firewall has different rules for incoming and outgoing packets. You should be able to forward incoming packets on port 4569 to your internal host and allow outgoing packets from 4569 tobe sent to the external hostcorrectly. How to do this depends on your firewall. Ifyoureallywanttochangetheport, well,youcouldlookatthefirstlineinthe[general]sectionofiax.confwhereitsaysport=,ortryreadingthedocsattheURLIpostedbelow:) Guills From: Liaan vd Merwe [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 01, 2005 1:17 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] IAX registration keep alives Hi already tried that After lots of troubleshooting, it seems like its a natting issue. do you know if there is a way to tell iax not to use 4569 as the from port? my firewall is getting VERY confused, for i publish that as well. thanks liaan - Original Message - From: Chamberland-Larose, Guillaume To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, February 01, 2005 9:42 PM Subject: RE: [Asterisk-Users] IAX registration keep alives I think qualify=1000 should work. Take a look at: http://www.voip-info.org/wiki-Asterisk+config+iax.conf Guills ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to make a call with asterisk from shell ? (orwith a .sh file )
Take a look at .call files: http://www.voip-info.org/wiki-Asterisk+auto-dial+out You should be able to write a script that generates call files pretty easily. Guillaume -Original Message- From: Mateo Meier [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 01, 2005 2:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how to make a call with asterisk from shell ? (orwith a .sh file ) Hello Guys Does anybody know how to make a call with asterisk with a shell comment ? I like to connect that comment with a small .sh script.. Is it actually possible to make a call with asterisk true shell or a .sh file ? Thank you for the help and regards from Switzerland Matt ___ 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] Audio Quality over LAN very bad
Maybe you're transcoding on the server with cpu intensive codecs? That would be the first thing I'd look at. Try using G.711 (ulaw)on both SIP phones and remove reinvite=no and canreinvite=no from your phone declarations in sip.conf. Hope that helps. Guills From: Nic le Roux [mailto:[EMAIL PROTECTED] Sent: Monday, January 31, 2005 7:01 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Audio Quality over LAN very bad Hi All, I'm running Asterisk on the following vendor_id : GenuineIntelmodel name : Celeron (Coppermine)cpu MHz : 668.202cache size : 128 KB with 192 MB Ram Audio coming from Asterisk (the demo ) is excellent when using a SIP phone on the LAN to Asterisk, and when dialling in from outside via ISDN to Asterisk. However, when connecting from SIP phone to SIP phone (across LAN) and dialling from externally to SIPwhich is on the local LAN it is very choppy and one can barely make out the other party. I'm using an Eicon Diva 2-m card and 100mb network all round. What could be the cause as I believe bandwidth is ruled out. Thanks and regards Nic ___ 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] FWD and IAX2
Hi, I had a FWD account set up with asterisk (using SIP) and it was working fine both ways. I switched to IAX2 and now I can't get incoming calls from FWD. People who call my FWD number get a 480 - user is not online message without any traffic reaching my box. I can call FWD numbers fine over IAX2. It seems fwd isn't trying to place the call over IAX2 because it thinks I'm not online. *CLI iax2 show registry Host UsernamePerceived Refresh State 65.39.205.121:4569xxx xxx.xxx.xxx.xxx:4569 60 Registered It looks like I'm registered though, and I can even call my own number fine. Other can't. :s Any suggestions? Anyone got something similar to work? Thanks, Guills ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_iax2.c problem?
-Original Message- From: Steve Kann [mailto:[EMAIL PROTECTED] Sent: Friday, January 28, 2005 12:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Adam Hart Subject: Re: [Asterisk-Users] chan_iax2.c problem? Looks like two bugs: 1) Apparently, asterisk's MOH is trying to send a zero-length voice frame out, (or maybe, it put data in the frame, but set samples to zero?). It could also be a different bug in MOH. I'll take a better look at this tonight and file a bug for MOH if I can find out what the issue is. 2) The line of code upon which chan_iax2 crashed, I think is one I wrote :). We should check for the zero case before trying to do this division. Can you file a bug on this, at the bugtracker? (2) should obviously be easy to fix as you saw. (1) I'm not sure about. I'll enter bug (2) in bugtracker right now. Also, I'm CC'ing Adam Hart so he knows about the issue in Firefly; he may want to make it more robust to this.. (it probably receives a voice frame with no data in this case, which isn't meaningful, but shouldn't crash things..). I should add, when asterisk crashes the way it did, Firefly thinks the call is still in progress, it will only hang up when I tell it to. Guills ___ 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] Tortoise CVS download for Asterisk Docs
- Install TortoiseCVS from the link provided - Inside the file explorer, right click where you want to check out the docs - click cvs checkout in the menu - in the CVSROOT field, enter :pserver:anonymous:@cvs.sourceforge.net:/cvsroot/asterisk (note the : before pserver AND after anonymous) - In Module, enter docs - Click OK I just installed TortoiseCVS to test this and it worked fine. If this doesn't work for you on your system, you can always install the command line cvs application that comes with cygwin (www.cygwin.com) and follow the command line instructions provided. This works even better when you want to sync docs and code all the time. Cheers, Guills From: dean collins [mailto:[EMAIL PROTECTED] Sent: Thursday, January 27, 2005 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Tortoise CVS download for Asterisk Docs http://www.asteriskdocs.org/modules/tinycontent/index.php?id=4 Can I make a suggestion that some documentation is provided for the Tortoise CVS download of the asterisk docs. I've tried every combination and I cant get it to work. I'm assuming it must work otherwise it wouldn't have been listed but for 60 seconds more work it would be a bigger benefit to the asterisk community. Cheers, Dean ___ 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] Need some advises configuring asterisk to callover INTERNET
Hi, You might want to first read http://www.digium.com/handbook-draft.pdf which explains most of the basic stuff. Most of the questions you'll have will be answered on http://www.voip-info.org or by reading http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_ v1/docs-html/book1.html Oh, and most questions have also been answered on this mailing list so look at http://lists.digium.com/pipermail/asterisk-users/ for the mailing list archives. When you've read that and understand what you're doing you should be well on your way :) Guills -Original Message- From: Ing. Ignacio Ortega A. [mailto:[EMAIL PROTECTED] Sent: Thursday, January 27, 2005 8:18 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Need some advises configuring asterisk to callover INTERNET Hello guys I just looking for some advises in order to configure my asterisk server to receive and make calls over internet, i got a 384 kb adsl connection. i just need any information regarding this matter , codecs (installing g729,g723) bandwidth, configuring public IP with adsl and others things to keep in mind, i need anything, so anyone that allready done this please take a few seconds to give me some advise Thank You ___ 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] Re: [Fwd: Re: [Asterisk-biz] bellster.net
Hi, In France, the second most important ADSL provider (named Free) offers a phone line (which uses VoIP but can only be used as a FXS) with unlimited free calls to landlines. I was wondering if I would use my Free phone line with Bellster as well, but I am not sure this is authorized by the ISP : http://adsl.free.fr/hd/cgv.html [in French] En particulier, l'utilisation du service à d'autres fins que privative (par exemple partage de l'accès téléphonique avec des personnes extérieures au foyer) ou raisonnable (taux d'utilisation manifestement excessif pour un abonné particulier par exemple) ainsi que l'utilisation à titre gratuit ou onéreux du service téléphonique de Freebox en tant que passerelle de réacheminement de communications, est strictement prohibée. In english, it is an extract of the ToS saying : the sharing of the line with people outside of the family, or the usage of the line as a communication bridge/gateway is strictly prohibited. Once this is said, if the number of calls done by Bellster users is limited, what's the probability the ISP discovers the trick...? It's an open question, I don't know how the ISP would react... Guillaume ___ 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] Question calling number
Hello all, I have a question concerning the calling number with an incoming PSTN call through a E100P : Here is what I see with a pri debug : Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '333007' ] [6c 0b 20 83 32 34 37 33 33 33 30 33 30] Calling Number (len=13) [ Ext: 0 TON: National Number (2) NPI: Unknown Number Plan (0) Presentation: Presentation allowed of network provided number (3) '33030' ] [70 05 81 34 32 34 33] There are two calling numbers : one is the real number (the first one), the second one is the group number (I don't know the correct word for this, maybe head number of the group ?). Asterisk is using the second one when I think it should use the first one. How could I tell asterisk to pick the first number ? The info is decoded by libpri, so it should be possible to change asterisk's default behaviour. Thanks a lot ! Guillaume ___ 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] X100P + Call-Waiting - Flash how-to.
Hi all I'm pretty sure someone must have done this before but I couldnt find any trace of it on the web so I thought I would drop a note about how I ended up doing it. I have also posted this info on voip-info. Warning : This is not very elegant and I'm currently trying to write a patch in order to make it better but so far, this the only way I've gotten this to work. Scenario : I have an asterisk box with an X100P card. When my phone line rings, it ring my SIP Phone (a Cisco 7940). I've got a call-waiting feature on my line and couldnt figure out how to trigger a flash in order to go from one call to another. Solution : 1st - The inbound context (in extensions.conf of course) [pstninbound] exten=s,1,Dial(SIP/cisco7940,40|Tt) exten=s,2,Voicemail(u1000) exten=s,3,Voicemail(b1000) The Tt option will allow you to transfer by hitting the # key. 2nd - The flash extension Now, somewhere in your extensions file, create a context that similar to this : exten=604,1,Flash() exten=604,2,Dial(SIP/cisco7940) By transfering a ZAP call to that extension, the line is flashed before ringing back to you. There are 2 ways to use this setup : Either by using the # key on an inbound call, or by using the BlndXfr key (if you do so, you can actually take the Tt our of the Dial sequence in your inbound context). When hearing the call-waiting tones, just blind transfer the call to extension 604. You ZAP channel will be flashed and it will ring back to you. Hope this helps. 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
