Re: [asterisk-users] Asterisk Radius CDR
Hi Ahmed and asterisk friends, So asterisk cannot contact the radius server. The radiusclient __can__ contact the radius server. Check in the asterisk log files why asterisk cannot contact the radius server! Be also aware of the user, who is running the daemons. This user might need read access to certain configuration files. On Wed, Sep 28, 2016 at 01:24:58PM -0400, Ahmed Munir wrote: > Hi Andrew and Willy, > > Thanks for sharing the info. > > As for enabling radius server debugging 'radiusd -X', made some test calls > don't see the radiusclient sending data to radius server. However, using > radtest or radiusclient testing, able to send data to radius server (after > enabling debug). > > For further testing, on my other server using OpenSIPs, setup the > radiusclient and data was able to send over to radius server without any > issue i.e. using same radiusclient config that I'm using for Asterisk > radiusclient. > > Btw, will try to work on Andrew advise and will update you if I make any > progress. > > > > Date: Wed, 28 Sep 2016 10:09:51 +0200 > > From: Willy Offermans > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > > > Subject: Re: [asterisk-users] Asterisk Radius CDR > > Message-ID: <20160928080951.ga4...@vpn.offrom.nl> > > Content-Type: text/plain; charset=us-ascii > > > > Hello Ahmed, Andrew, and asterisk friends, > > > > Some time ago, I ran into similar problems as well :) I can confirm the > > statement of Andrew: Turn on the logging facilities and you will find your > > issue most likely. However, you need also a strategy. ``Radius client > > testing'' as you mentioned, can mean anything. The point is, can asterisk > > talk to the freeradius server via the client settings? To my opinion, this > > is easy to test. Maybe the message: ``cdr_radius.c:208 radius_log: Unable > > to create RADIUS record. CDR not recorded'' already implies that this is > > not possible. I cannot judge it. You can by turning on radiusd -X and have > > a close look to the output. > > > > On Wed, Sep 28, 2016 at 07:59:13AM +0800, Andrew Ivins wrote: > > > Hi Ahmed, > > > > > > I ran into similar problems. freeradius-client returns the same error > > code > > > for numerous failure cases, so Asterisk doesn't get an opportunity to log > > > anything useful. If you look here: > > > > > > https://github.com/FreeRADIUS/freeradius-client/blob/master/ > > lib/buildreq.c > > > > > > You'll see many instances where it returns ERROR_RC. You are almost > > > certainly running into one of these. I ended up putting in print debug > > into > > > that file and recompiling. I think in my case it was as simple as a > > > hostname not resolving. Once you're not working blind, you'll find what > > is > > > happening pretty quickly. > > > > > > Andrew > > > > > > On 28 September 2016 at 03:32, Ahmed Munir > > wrote: > > > > > > > I did radius client status testing with radius server, able to access > > the > > > > radius server. However, still getting radius CDR issue after setting > > debug > > > > level 8 even granting 666 access to radiusclient-ng config files. > > > > > > > > message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. > > CDR > > > > not recorded! > > > > > > > > Please advise if I missed out anything. > > > > > > > > > > > > Date: Mon, 26 Sep 2016 12:09:34 +0200 > > > >> From: Willy Offermans > > > >> To: Asterisk Users Mailing List - Non-Commercial Discussion > > > >> > > > >> Subject: Re: [asterisk-users] Asterisk Radius CDR > > > >> Message-ID: <20160926100934.gb4...@vpn.offrom.nl> > > > >> Content-Type: text/plain; charset=us-ascii > > > >> > > > >> > > > >> Hello Ahmed, > > > >> > > > >> On Fri, Sep 23, 2016 at 04:12:42PM -0400, Ahmed Munir wrote: > > > >> > Hi, > > > >> > > > > >> > I've recently setup Asterisk with Radius CDR by following the > > document: > > > >> > https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend. > > > >> > > > > >> > The issue currently I'm facing is after turning on the debug getting > > > >> > message: cdr_radius
Re: [asterisk-users] Asterisk Radius CDR
Hello Ahmed, Andrew, and asterisk friends, Some time ago, I ran into similar problems as well :) I can confirm the statement of Andrew: Turn on the logging facilities and you will find your issue most likely. However, you need also a strategy. ``Radius client testing'' as you mentioned, can mean anything. The point is, can asterisk talk to the freeradius server via the client settings? To my opinion, this is easy to test. Maybe the message: ``cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR not recorded'' already implies that this is not possible. I cannot judge it. You can by turning on radiusd -X and have a close look to the output. On Wed, Sep 28, 2016 at 07:59:13AM +0800, Andrew Ivins wrote: > Hi Ahmed, > > I ran into similar problems. freeradius-client returns the same error code > for numerous failure cases, so Asterisk doesn't get an opportunity to log > anything useful. If you look here: > > https://github.com/FreeRADIUS/freeradius-client/blob/master/lib/buildreq.c > > You'll see many instances where it returns ERROR_RC. You are almost > certainly running into one of these. I ended up putting in print debug into > that file and recompiling. I think in my case it was as simple as a > hostname not resolving. Once you're not working blind, you'll find what is > happening pretty quickly. > > Andrew > > On 28 September 2016 at 03:32, Ahmed Munir wrote: > > > I did radius client status testing with radius server, able to access the > > radius server. However, still getting radius CDR issue after setting debug > > level 8 even granting 666 access to radiusclient-ng config files. > > > > message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR > > not recorded! > > > > Please advise if I missed out anything. > > > > > > Date: Mon, 26 Sep 2016 12:09:34 +0200 > >> From: Willy Offermans > >> To: Asterisk Users Mailing List - Non-Commercial Discussion > >> > >> Subject: Re: [asterisk-users] Asterisk Radius CDR > >> Message-ID: <20160926100934.gb4...@vpn.offrom.nl> > >> Content-Type: text/plain; charset=us-ascii > >> > >> > >> Hello Ahmed, > >> > >> On Fri, Sep 23, 2016 at 04:12:42PM -0400, Ahmed Munir wrote: > >> > Hi, > >> > > >> > I've recently setup Asterisk with Radius CDR by following the document: > >> > https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend. > >> > > >> > The issue currently I'm facing is after turning on the debug getting > >> > message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. > >> CDR > >> > not recorded! > >> > > >> > I've checked and grant access 666 to radiusclient config files: servers > >> & > >> > dictionary.digium and 777 to '/var/run/radius.seq'. I've noticed that > >> > /var/run/radius.seq is not getting updated. > >> > > >> > > >> > Further added, in asterisk CLI while running command: cdr show status > >> > getting results below; > >> > > >> > Call Detail Record (CDR) settings > >> > -- > >> > Logging:Enabled > >> > Mode: Simple > >> > Log unanswered calls: No > >> > Log congestion: No > >> > > >> > * Registered Backends > >> > --- > >> > cdr-syslog > >> > Adaptive ODBC > >> > cdr-custom > >> > csv > >> > radius > >> > > >> > > >> > Please advise if I may missed any steps. > >> > > >> > -- > >> > Regards, > >> > > >> > Ahmed Munir Chohan > >> > >> I cannot advice you about steps you might have missed, probably none. To > >> my > >> experience, the documentation is not sufficient. > >> > >> I can tell you that freeradius can be run in debug mode: radiusd -X Do > >> this > >> and have a close look to the output. > >> > >> If you cannot find any attempt to connect to the freeradius server you > >> need > >> to have a close look to the asterisk log files as well. Figure out what is > >> going wrong. There should be some clue. > >> > >> I don't understand the grant access settings. Figure out the user which is > >> running asterisk and
Re: [asterisk-users] Asterisk Radius CDR
Hello Ahmed, On Fri, Sep 23, 2016 at 04:12:42PM -0400, Ahmed Munir wrote: > Hi, > > I've recently setup Asterisk with Radius CDR by following the document: > https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend. > > The issue currently I'm facing is after turning on the debug getting > message: cdr_radius.c:208 radius_log: Unable to create RADIUS record. CDR > not recorded! > > I've checked and grant access 666 to radiusclient config files: servers & > dictionary.digium and 777 to '/var/run/radius.seq'. I've noticed that > /var/run/radius.seq is not getting updated. > > > Further added, in asterisk CLI while running command: cdr show status > getting results below; > > Call Detail Record (CDR) settings > -- > Logging:Enabled > Mode: Simple > Log unanswered calls: No > Log congestion: No > > * Registered Backends > --- > cdr-syslog > Adaptive ODBC > cdr-custom > csv > radius > > > Please advise if I may missed any steps. > > -- > Regards, > > Ahmed Munir Chohan I cannot advice you about steps you might have missed, probably none. To my experience, the documentation is not sufficient. I can tell you that freeradius can be run in debug mode: radiusd -X Do this and have a close look to the output. If you cannot find any attempt to connect to the freeradius server you need to have a close look to the asterisk log files as well. Figure out what is going wrong. There should be some clue. I don't understand the grant access settings. Figure out the user which is running asterisk and set the setting appropriately! I remember that I needed the following access setting: -rw-r- 1 root asterisk /usr/local/etc/radiusclient-ng/servers So read access for asterisk to the servers file. This was not documented at all, but somehow logical, if you figured it out. -- Met vriendelijke groeten, With kind regards, Mit freundlichen Gruessen, De jrus wah, Wiel * W.K. Offermans Powered by (__) \\\'',) \/ \ ^ .\._/_) www.FreeBSD.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Multiple phones when one is unregistered
Dear D'Arcy J.M. Cain and asterisk friends, On Tue, Aug 30, 2016 at 09:56:05AM -0400, D'Arcy J.M. Cain wrote: > I have an extension that looks like this: > > exten => 55,1,Verbose(Door buzzer calling) > same => n,Dial(SIP/user1&SIP/user2&SIP/user3) > > The idea is that any of the three users can answer the phone to let > someone in. The problem is that if, say, user2 unplugs his phone then > the call immediately goes to his voice mail and the other two do not > have the ability to open the door. > > Is there any way to direct only to phones in a list that are currently > registered? I am sure that I can write a rather convoluted extension > to check for registrations and create a dial command but I am hoping > that there is an easier way so that I can create these types of > extensions for other clients easily as well as being able to add and > remove destinations quickly. > To my opinion, you need a queue with dynamic agents. SIP/userX subscribes and checkouts manually, or better SIP/userX adds to the specific queue at the moment she or he registers to the system. I know how to do the former, but I would like to know about the latter. If someone could comment on that, I would highly appreciate. -- Met vriendelijke groeten, With kind regards, Mit freundlichen Gruessen, De jrus wah, Will * W.K. Offermans Powered by (__) \\\'',) \/ \ ^ .\._/_) www.FreeBSD.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016 http://www.asterisk.org/community/astricon-user-conference New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13 with LDAP ? (single sign on )
Hello Kevin, hello asterisk friends, On Sat, Jun 11, 2016 at 05:33:54AM +, Kevin Long wrote: > > > Is it possible to configure Asterisk such that numerical extensions and/or > usernames, would be populated from LDAP, as well as authenticate the > endpoints where the “SIP secret” is equal to the user’s hashed password in > LDAP? > > > I’d like to use LDAP for single-signon as I do with a number of other > applications, and am curious if anyone has a working example or if this is > even possible? > > > Thank you, > > Kevin Long > I'm puzzling with a somehow similar problem. I like to couple asterisk's authentication, authorisation and accounting with a radius server. The radius server will use a ldap server as database for passwords and other data. The real benefit of this setup is that a ldap database is not designed for authentication, it is a kind of database. A radius server is designed for authentication. If I understand it correctly then SIP authentication works with HTTP digest authentication, a challenge response mechanism. A ldap database does not know what to do with this mechanism. It cannot deal with authentication mechanisms. A radius server, such as freeradius, can handle this mechanism of authentication. It is designed for this. I'm looking for info on how to setup this up: asterisk <--> freeradius <--> openldap and already asked for info or documentation on this list. However without any response so far. I also asked if asterisk supports pam for authentication. Also this question was not answered so far. Another strategy can be to use the ldap server to record all necessary data and asterisk to retrieve this data from the ldap database. With other words and have a look to https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver sippeers = ldap,"ou=sip,dc=example,dc=domain",sip sipusers = ldap,"ou=sip,dc=example,dc=domain",sip extensions = ldap,"ou=extensions,dc=example,dc=domain",extensions Asterisk will then deal with authentication, authorisation and accounting. This is how you imagined to set it up, if I understand it correctly. However, if you look at it from a distance and in detail, then asterisk should not concentrate on designing to handle this. A radius server can be involved for this work. Asterisk could then concentrate on its core business and that is managing voice and voice/video connections. The radius server does what it good at is: authentication, authorisation and accounting. I guess that most commercial implementations use something like asterisk <--> radius <--> database for authentication, authorisation and accounting. However, the underlying information on how to set this up is not willingly shared. If I cannot get more details on asterisk <--> freeradius <--> openldap, I will spent the next days to look in more detail to https://wiki.asterisk.org/wiki/display/AST/LDAP+Realtime+Driver I can keep you updated, if you are interested. -- Met vriendelijke groeten, With kind regards, Mit freundlichen Gruessen, De jrus wah, Will * W.K. Offermans Powered by (__) \\\'',) \/ \ ^ .\._/_) www.FreeBSD.org -- _ -- 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
[asterisk-users] asterisk pam authentication support
Dear asterisk friends, Can someone tell me whether asterisk supports PAM authentication or not? -- Met vriendelijke groeten, With kind regards, Mit freundlichen Gruessen, Will * W.K. Offermans Powered by (__) \\\'',) \/ \ ^ .\._/_) www.FreeBSD.org -- _ -- 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
[asterisk-users] asterisk and freeradius AAA
Dear asterisk friends, I like to use asterisk and to do authentication, authorization and accounting (AAA) for it with freeradius. I looked for any documentation on the net, but could not find much useful and detailed information. I have made a first shot with radiusclient-ng. I configured cdr.conf, cel.conf and radiusclient.conf. Execution of radexample gave a positive result. So at least for authentication, the setup of radiusclient.conf seems to work. However asterisk throws following error upon a phonecall: [Jun 8 12:17:29] ERROR[101041]: cdr_radius.c:228 radius_log: Failed to record Radius CDR record! The debug of freeradius does not show any activity. Is it possible to combine asterisk and freeradius for AAA? If yes, is there any documentation available and can you point me to it? Can I debug the error message ERROR[101041] in more detail. -- Met vriendelijke groeten, With kind regards, Mit freundlichen Gruessen, Will * W.K. Offermans Powered by (__) \\\'',) \/ \ ^ .\._/_) www.FreeBSD.org -- _ -- 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
[asterisk-users] Caller ID on Channelized T1 (E&M Wink)
Hi, Normally my T1 implementations are PRI. However, I do have a customer who uses channelized T1 (24 channels). I have setup a 'test' environment, and have two T1 channels back-to-back in my [*] box. Both are setup with signalling => em_w. Calls DO go back & forth, but I can not see the callerID being passed. Any ideas? WW -- Willy Wouters, PhD Asterisk Telephony Web Applications MAGU ENTERPRISES Tel: 713-474-1534 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Asterisk call quality detection
Henry, I have never seen VNAK in any of the asterisk logs. What version of asterisk are you running? Henry Cobb wrote: grep VNAK /var/log/asterisk/full | cut -d ' ' -f 4 | cut -d : -f 1 | uniq -c Needs a bit of an adjustment between the 1-9th and 10th-31st of the month so I'm looking for something to chomp this automatically. -- Willy Wouters, PhD Asterisk Telephony Web Applications MAGU ENTERPRISES Tel: 713 474-1534 Fax: 501 665-1544 ___ --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] How does Norvergence do it ?
So a guy shows up at the the office, after making an appointment with the office manager / receptionist to talk 'phone systems'. After her eyes glaze over, with him talking T1 and Frame-Relay I get to see him. He's from Norvergence. Well dressed. Tells me they can do a T1 for $79, with unlimited local & long distance for free. It also does 'internet'. 'Just give me copies of your phone bill'. I ask some questions, like number porting, like provisioning of DID numbers, like CIR on the data etc. Now HIS eyes glaze over. That's technical talk ... He's just there to follow up on the appointment and 'qualify' the customer to see if we are worthy of their cheap service. After I looked at their website, I can hear 'quack quack'. Cheers, WW Willy Wouters ypOne Publishing ___ 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] iconnecthere behind NAT, strange deal
> Is anyone on the list successfully using iconnecthere > behind NAT? Yes (unless it broke during the last 12 hours). I gave my daughter a SIP phone and an Iconnecthere account. I have successfully used Sipura 2000, a Grandstream and a Cisco ATA box with that account, and yes, she is behind a *standard* LinkSys NAT router which itself gets a dynamic WAN address from the service provider. Now, that being said, I am not totally happy with Iconecthere. Their service availability has been spotty, and they can not seem to be able to forward the correct (outgoing) callerID. > I was, for over a year, and then I changed my "plan" with > them. Now all my calls get intercepted immediately, What 'plan' change did you make? > FWIW I used to prepend "" to the dialed number, and it > worked fine until last week. I don't recall having to do that. Iconnecthere just gives us (USA based) international dialtone. Willy - Original Message Follows - > I've been to the WIKI and I've searched the archives. > > Is anyone on the list successfully using iconnecthere > behind NAT? > > I was, for over a year, and then I changed my "plan" with > them. Now all my calls get intercepted immediately, > "We're sorry, but your account is temporarily > unavailable." > > Incoming calls work just fine. > > I contacted their so-called "customer care," which has > sent me repeated replies asking me to give them the > version of my PC phone. When I say I don't have one, > they say, "Sorry, we only help those who do." > > I like to play with their GSM stuff, so I hate to let the > account go, but if no one here knows what might be going > on, they certainly don't. > > FWIW I used to prepend "" to the dialed number, and it > worked fine until last week. > > Thx. > > B. > ___ > 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 Willy Wouters ypOne Publishing ___ 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] PRI: This number has been disconnected
THANK YOU - Original Message Follows - > Hi, > > it seems like you are using the 'r' option of app_dial. > This will fake ring indication and will not pass any audio > until the call is answered. THANK YOU! That was it! As usual as was looking for a complex solution instead of seeing the obvious. I must admit that with the recent issues with the 'r' option I was confused about its meaning. I remember that some (short) time ago, without an 'r' no ringing would be heard at all. This must have been a bug. However, that is when I added 'r' to all my Dial commands. Thanks again, Willy > What does your dial extension look like? > > best regards > > kapejod > -- > Klaus-Peter Junghanns > > CEO, CTO > Junghanns.NET GmbH > Breite Strasse 13a - 12167 Berlin - Germany > fon: (de) +49 30 79705390 > fon: (uk) +44 870 1244692 > fax: (de) +49 30 79705391 > iaxtel: 1-700-157-8753 > http://www.Junghanns.NET/asterisk/ > > > Am So, 2004-04-18 um 17.09 schrieb [EMAIL PROTECTED]: > > All, > > When calling an invalid number using, I expect to hear: > > "dooh-deeh-daah We're sorry you have reached a number > > which has been disconnected ..." > > And that is indeed what I hear when I dial out from [*] > > using analog FXO, or VoicePulse or NuPhone. When I dial > > that same number trough the T1 / PRI interface however, > > I continually hear ringing, and then the call gets > > hungup. Any ideas anyone? > > It kinda annoys our users, since they like to *know* > > when they dial an invalid number. > > TIA, > > WW > > > > Willy Wouters > > ypOne Publishing > > > > ___ > > 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 Willy Wouters ypOne Publishing ___ 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] PRI: This number has been disconnected
- Original Message Follows - > All, > When calling an invalid number using, I expect to hear: > "dooh-deeh-daah We're sorry you have reached a number > which has been disconnected ..." > And that is indeed what I hear when I dial out from [*] > using analog FXO, or VoicePulse or NuPhone. When I dial > that same number trough the T1 / PRI interface however, I > continually hear ringing, and then the call gets hungup. Further info ... So I ran the scenario with PRI DEBUG SPAN 1 and here is what I can see: ** Call to a Good Number ** < Message type: CALL PROCEEDING (2) < Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 <ChanSel: Reserved < Ext: 1 Coding: 0 Number Specified Channel Type: 3 < Ext: 1 Channel: 2 ] -- Processing IE 24 (Channel Identification) < Protocol Discriminator: Q.931 (8) len=9 < Call Ref: len= 2 (reference 32781/0x800D) (Terminator) < Message type: ALERTING (1) < Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) < Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] etc etc etc ** Call to a Bad Number ** < Message type: CALL PROCEEDING (2) < Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 <ChanSel: Reserved < Ext: 1 Coding: 0 Number Specified Channel Type: 3 < Ext: 1 Channel: 2 ] -- Processing IE 24 (Channel Identification) < Protocol Discriminator: Q.931 (8) len=13 < Call Ref: len= 2 (reference 32782/0x800E) (Terminator) < Message type: PROGRESS (3) < Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) < Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] < Progress Indicator (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) < Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (Cause) -- Processing IE 30 (Progress Indicator) < Protocol Discriminator: Q.931 (8) len=9 < Call Ref: len= 2 (reference 32782/0x800E) (Terminator) < Message type: DISCONNECT (69) < Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the remote user (4) < Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (Cause) -- Channel 2, span 1 got hangup As you can see, the PRI reports different stuff. It also has somewhere a 'Unallocated (unassigned) numnber (1). Then the PRI proceeds to send 'Inband information' which presumably is the recording I am never hearing. > Any ideas anyone? > It kinda annoys our users, since they like to *know* when > they dial an invalid number. > TIA, > WW > > Willy Wouters > ypOne Publishing > > ___ > 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 Willy Wouters ypOne Publishing ___ 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] PRI: This number has been disconnected
All, When calling an invalid number using, I expect to hear: "dooh-deeh-daah We're sorry you have reached a number which has been disconnected ..." And that is indeed what I hear when I dial out from [*] using analog FXO, or VoicePulse or NuPhone. When I dial that same number trough the T1 / PRI interface however, I continually hear ringing, and then the call gets hungup. Any ideas anyone? It kinda annoys our users, since they like to *know* when they dial an invalid number. TIA, WW Willy Wouters ypOne Publishing ___ 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] Strange T1 Problem
Lookie here:
This is what you have
> exten => 1234567890,2,Dial(SIP/user1|r)
But, perhaps, here's what it shouls be:
exten => 1234567890,2,Dial(SIP/user1||r)
The second argument is *timeout*.
Normally you'd have something like
Dial(Channel,time,options)
exten => 1234567890,2,Dial(SIP/user1|60|r)
But the empty time works as well. It will just ring
forever.
Cheers,
Willy
- Original Message Follows -
>
> On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
>
> > Explicitly answer the line. If that doesn't handle
> > inband audio, there is a r flag to dial. This was
> discussed very recently.
>
> This must be a different problem, because neither of those
> solutions worked.
>
>
>
> zapata.conf sends call to fixup context:
>
>
> [fixup]
>
> ; Receive call as **
> exten => _.,1,Answer
> exten => _.,2,Cut(CALLING=EXTEN,*,2)
> exten => _.,3,SetCIDNum(${CALLING})
> exten => _.,4,Cut(CALLED=EXTEN,*,3)
> exten => _.,5,Goto(default|${CALLED}|1)
>
>
> [default]
>
> exten => 1234567890,1,Answer
> exten => 1234567890,2,Dial(SIP/user1|r)
>
>
> user1's phone rings, but no ring from PSTN caller. user1
> picks up, both can talk ok.
>
>
> I have been using cvs stable branch. I will try HEAD and
> see if that fixes it as suggested by Eric.
>
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RE: [Asterisk-Users] Hunting S(n)IPs
> Another thing to try is to disable call waiting on the > [EMAIL PROTECTED] phone (if call waiting is enabled, it's doing > what you've asked it to)... > Yep, except on the Polycom, we have found no way to disable call-waiting. WW Willy Wouters ypOne Publishing ___ 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] Hunting S(n)IPs
> 2) I think you should have been looking at incominglimit, > not outgoinglimit, or possibly both of them together in > some combination. > Another perspective issue. Apparantly 'incoming' means into the [*] box, and outgoing is leaving the [*]. In any case, I tried both, but 'outgoing' is confirmed broken. WW Willy Wouters ypOne Publishing ___ 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] T100P Timing Was:T100P/ ZAP / PRI errors
Don & others, Thank you for your answer. The fog maybe lifting ;). The zaptel.conf file has the following in its comments: # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of "1". For a secondary, use "2", and so on. # To not use this as a sync source, just use "0" # You stated: > > In this case you want to use loop timing (use the T1 as a > sync source) so, > > span=1,1,0,esf,b8zs > I have no reason to doubt what you wrote, so I already changed the timing parameter for my system ;). I did have it set as span=1,0,0, ... Now, please, in what scenario would one select option '0' and win what scenario would one use option '2'? I guess what gets me confused is the terminology 'primary sync source' and 'span'. The way I read it is span === digium card. If we are taking timing from the Telco, then the digium card should slave from the Telco and cannot therefore be a primary sync source (for the loop), so I had it set at '0'. After your explanation, it now sounds like we are talking about timing for the BOX, not the loop. Therefore, the LOOP is the Primary Source for the BOX, hence use '1'. If we are generating the loop (fxs role) then there is no Telco to slave off, and the BOX should take its timing from the CARD hence select '0'. If this reasoning is correct, then when would one reasonably use '2' and even '3' etc. Sorry to be a pest about this. WW Willy Wouters ypOne Publishing ___ 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] T100P / ZAP / PRI errors
Now you've got me utterly confused ... So, in layman's terms, if I connect a T100P to a circuit provided by the Telco, and the Telco says that they will provide timing, I have to put WHAT? span=1,0,0,esf,b8zs,yellow this means '0' this span is not a sync source, i.e. the Telco will provide my 8kHz. Could one use '2' with impunity (span=1,2,0,...)? I am still not clear under which circumstances one should use '0' versus '2'. WW - Original Message Follows - > Holy crap people, trim your replies! > > > You didn't say what's at the other end of your PRI line, > > but you might try having the other end be the timing > > sync source. Try: span=1,0,0,esf,b8zs instead. Maybe > that will help. > > We need to get this documented *clearly* once and for all. > > Zaptel T1/E1 hardware either free-runs to its own internal > 8kHz time source, or it tries to lock to the recovered > clock from the line. > > Zapata.conf says that timing of 0 means "do not use this > span for timing." Zero does not mean "slave timing", it > means not to use this span as a recovered clock source > for timing at all. Timing values of 1 or 2 mean try to > lock the internal clock to the recovered clock from the > span. > > A value of 0 means that this span's recovered clock never > gets used as a timing source. A value of 1 means that > this span is the primary clock source -- If the span is > up, try to lock the internal clock to the clock recovered > from this span. A value of 2 means to use this span for > timing only if the primary span is down. > > To reiterate: a value of 0 means that the other end must > be locking to the zaptel's clock or else clock slips will > occur. > > Feel free to correct me if I'm wrong, but I am pretty sure > I have this right. :-) > > Regards, > Andrew > ___ > 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 Willy Wouters ypOne Publishing ___ 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 NTL (ex Cable + Wireless)
X100P hangup detection works only sporadically in the US as well. Probably a bug in hardware and/or software. Not sure. It just does not work for a lot of people. WW - Original Message Follows - > Firstly, let me just say I am new to asterisk and if > anything I've said is covered in an FAQ or in previous > posts I apologise but I have tried searching and I've > attempted a few of the things I found but they didn't > help. > > Has anybody got any experience using an X100P on an NTL > phone line in the UK (I'm in an ex Cable & Wireless area > if that makes any difference). > > The problem I'm having (and judging by the number of > references to it I've found searching it is a common one) > is getting * to detect when the line has been hung up. > It doesn't matter if it comes through to a person > directly as they can just hang that phone up, but when it > hits voicemail, and it sits there for two minutes > recording an empty message, and then emails it to the > person it can be a bit annoying! > > There are a couple of possible things that I realize could > be causing it, the most major being that the phone wiring > to the port that asterisk is plugged into is a bit dodgy > and I think there is a fair amount of interference which > I suspect could be annoying it a bit. > > The other possible problem is that when someone hangs up, > the tone that comes down the phone line to indicate that > it has been hung up lasts for about 4 seconds, and then > the line just goes silent - would this mean that the busy > detect mechanism would not detect that it has been hung > up? > > Thanks in advance, > Alex Brett > [EMAIL PROTECTED] > +44 (0)870 744 2170 > http://www.loho.co.uk/ > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Hunting S(n)IPs
Another observation of something which doesn't work:
exten => 3200,1,Dial(SIP/3200,20,tTr)
exten => 3200,2,Playback(tt-weasels)
exten => 3200,3,Hangup
exten => 3200,102,Dial(SIP/3201,20,tTr)
exten => 3200,103,Playback(tt-weasels)
exten => 3200,104,Hangup
exten => 3200,203,Dial(SIP/3202,20,tTr)
exten => 3200,204,Playback(tt-weasels)
exten => 3200,205,Hangup
The [EMAIL PROTECTED] phone does NOT give a BUSY indication even afer
the first call has been answered. Therefore, Call#2 happily
dials 3200 again, although 3200 is currently talking. I also
tried to limit the number of calls going to the phone with
outgoinglimit=1 in the sip.conf, but that makes no
difference either. According to the wiki that functionality
is broken.
- Original Message Follows -
> Hi Akk,
> If this has been discussed/done then apologies be-4-hand.
> I did not find it in the Wiki or the Archives. Here's the
> question.
> We have incoming PRI lines, all on the same main number.
> An attendant is supposed to handle all incoming calls.
> Now, let's say I have a multi-line SIP phone. For
> argument's sake (and to keep it simple) say I only have
> two lines. We'll call them SIP/att-0 and SIP/att-1. Here's
> the desired behavior:
> Call comes in. Gets to Dial(SIP/att-0)
> Other call comes in b4 first one is answered. Gets to
> Dial(SIP/att-1)
> Or, if Line-0 is busy (however) I still want to ring
> line-1. Kinda-like a hunt group. The problem I am having
> is that I cannot find out (real-time - in the dial plan)
> whether a particular channel is already in use. Otherwise
> a GotIf() might do the trick. I tried to set a parameter
> in the DB to indicate that a chan is in use, e.g.
> exten => s,1,DBPut(inuse/chan${ARG1}=1)
> exten => s,2,Dial(SIP/att-${ARG1})
> however, I do not seem to be able to catch the event wich
> releases the channel in order to reset the DB variable.
> exten => h,1,DCPut(inuse/chan${ARG1}=0) ; this never gets
> executed
>
> Any ideas?
> Cheers,
> WW
>
> Willy Wouters
> ypOne Publishing
>
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ypOne Publishing
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[Asterisk-Users] Hunting S(n)IPs
Hi Akk,
If this has been discussed/done then apologies be-4-hand. I
did not find it in the Wiki or the Archives. Here's the
question.
We have incoming PRI lines, all on the same main number. An
attendant is supposed to handle all incoming calls. Now,
let's say I have a multi-line SIP phone. For argument's
sake (and to keep it simple) say I only have two lines.
We'll call them SIP/att-0 and SIP/att-1. Here's the desired
behavior:
Call comes in. Gets to Dial(SIP/att-0)
Other call comes in b4 first one is answered. Gets to
Dial(SIP/att-1)
Or, if Line-0 is busy (however) I still want to ring line-1.
Kinda-like a hunt group. The problem I am having is that I
cannot find out (real-time - in the dial plan) whether a
particular channel is already in use. Otherwise a GotIf()
might do the trick. I tried to set a parameter in the DB to
indicate that a chan is in use, e.g.
exten => s,1,DBPut(inuse/chan${ARG1}=1)
exten => s,2,Dial(SIP/att-${ARG1})
however, I do not seem to be able to catch the event wich
releases the channel in order to reset the DB variable.
exten => h,1,DCPut(inuse/chan${ARG1}=0) ; this never gets
executed
Any ideas?
Cheers,
WW
Willy Wouters
ypOne Publishing
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Re: [Asterisk-Users] Booting error - Unable to specify channel 2: No such device
Hi .. This may seem odd, but this problem is reminescent of the troubel I had when first starting [*] on my test setup. Two suggestions. (a) on loading the modules make sure you explicitely load in the following order (forget modprobe) insmod zaptel; insmod wcfxo; insmod wcfxs; Do a ztcfg -vvv to confirm & start [*]. (b) in my zapata.conf, I foumd it useful to put group=1 on top for the FXO channel and group=2 for FXS, i.e. see below: ; ; Zapata telephony interface ; Configuration file [channels] context=inbound-calls language=en musiconhold=default group=1 signalling=fxs_ks echocancel = 64 echocancelwhenbridged = no echotraining=yes rxgain => 20% txgain => -5% channel => 1 group=2 echocancel = no signalling=fxo_ks mailbox = 2100 channel => 2 -- Now it works every time. Hope this helps Willy - Original Message Follows - > On Mon, 2004-04-12 at 03:39, Anon wrote: > > On Sunday 11 April 2004 07:18 pm, Todd Lieberman wrote: > > > do a cat /proc/interupts > > > > > > your should see your hardware showup. > > OK... > > cat /proc/interrupts > >CPU0 > > 0: 494600 XT-PIC timer > > 1: 5588 XT-PIC keyboard > > 2: 0 XT-PIC cascade > > 8: 1 XT-PIC rtc > > 9: 266296 XT-PIC ehci_hcd, es1371 > > 10:4892925 XT-PIC usb-uhci, wctdm > > 11:4936093 XT-PIC ide1, usb-uhci, > > usb-uhci, wcfxo, eth0 12: 108609 XT-PIC > > PS/2 Mouse 14: 22056 XT-PIC ide2 > > 15: 83 XT-PIC ide3 > > NMI: 0 > > LOC: 494575 > > ERR: 1278 > > MIS: 0 > > > > I don't see wcfxs. I did "modprobe wcfxs" and "cat > > /proc/interrupts" still shows the same output. What do > > you think is causing to not show on the list above? > > It is there it's hiding under the name of wctdm, but at > the moment I'd say that's the least of your problems. > Interrupts 10 and 11 both have other things sharing the > interrupts with Digium cards, most people will tell you > that that's not good. Interrupt 11 looks horrible. Having > been through this hell recently I can only suggest > repositioning cards in different slots. Interestingly > where is ide0 and do you really need ide2/3? > > -- > Dave Cotton > Directeur > Linux Autrement > 193 rue Marcel Cerdan > 84270 Vedene > 04 90 23 30 81 > <http://www.linuxautrement.com> > IAX 17004902330 FWD 42651 > > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Zaptel/PRI problem
PIC ide2 > > > 15: 967550 XT-PIC t1xxp > > > NMI: 1 > > > ERR: 0 > > > > > > Any ideas? > > > Tan > > > > > > > > > -Original Message- > > > From: [EMAIL PROTECTED] > > > [mailto:[EMAIL PROTECTED] On > > > Behalf Of [EMAIL PROTECTED] > > > Sent: 10 April 2004 17:23 > > > To: [EMAIL PROTECTED] > > > Subject: Re: [Asterisk-Users] Zaptel/PRI problem > > > > > > > > > To the list ... > > > The problem appears to be fixed. Short answer: > > > interrupts. When looking at cat /proc/interrupts it > > > was seen that the wct1xxp card was sharing interrupts > > > with a.o. the eth0 main ethernet driver. The solution > > > we simple: move the T1 card to another PCI slot. Upon > > > checking cat /proc/interrupts again, the card now had > > > its very own interrupt. So far I have NOT seen any > > > PRI notices or warnings. User experience has been > > limited due to a light workload on Friday, and, of > > > > > course, this being Easter weekend. Since this is such > > > a recurring problem (from witnessing list postings), > > > perhaps digium should have a notice or something about > > > this when they ship the cards out. Especially a > > > relative newbie (first T1 on an asterisk box) like me > > > is likely to even look at this interrupts situation. > > > Especially when there are other issues (zaptel.conf > > > and zapata.conf) to get a functional system. > > > Maybe this deserves a wiki input. In any case, > > > Happy Easter > > > WW > > > > > > - Original Message Follows - > > > > Dimitri, > > > > I just got off the phone with digium. Here's what I > > > > (from my notes) the event codes mean > > > > Event 4: Alarm detected > > > > Event 5: Alarm cleared > > > > Event 6: Abort HDLC Frams > > > > Event 8: Bad HCS > > > > The 6 & 8 which occur sporadically are possibly > > > > causing the observed symptoms. > > > > Now ... what causes 6 & 8 is the question. > > > > Interrupt conflicts was one suggested possibility. > > > > Another possibility is 'stuff' from the Telco which > > > > is not understood / mis-understood by the driver. > > > > I'll keep the list posted. > > > > Willy > > > > > > > > - Original Message Follows - > > > > > Dear Willy > > > > > i notice the same problem with my E100P using > > > > > the latest cvs zaptel driver i have try every > > type of config in /etc/zaptel.conf > > > > > > > to check if i have missed something in timing conf > > but nothing... > > > > > > > Digium help... :-) thanks in advance Dimitri > > > > > > > > > > On Thursday 08 April 2004 23:07, > > > > > > [EMAIL PROTECTED] wrote: Chris, > > > > > > Thank you for posting this. Since it concerns > > > > > > my 'production' system, let me comment. After > > > > > > 'downshifting' to a previous release (for no > > good reason other than desperation and teh fact > > > > > > > > that an earlier list entry had commented that it > > > > > > cleared up the problems) I am sad to report that > > > > > > the system failed again. Miscellaneous > > > > > > throughout the day: Apr 8 13:41:27 > > > > > > WARNING[-1210639440]: PRI: Read on 32 failed: > > > > > > Unknown error 500 Apr 8 13:41:27 > > > > > > NOTICE[-1210639440]: PRI got event: 8 on span 1 > > > > > > Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read > > > > > > on 32 failed: Unknown error 500 Apr 8 13:41:27 > > > > > > NOTICE[-1210639440]: PRI got event: 6 on span 1 > > > > > > Apr 8 13:42:07 WARNING[-1210639440]: PRI: Read > > > > > > on 32 failed: Unknown error 500 > > > > > > Apr 8 13:42:07 NOTICE[-1210639440]: PRI got > > > > > > event: 8 on span 1 > > > > > > Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read > > > > > > on 34 failed: Unknown error 500 > > > > > > Apr 8 16:44:01 NOTICE[-1210631248]: PRI got > > > > > > event: 6 on span 1 > > &g
RE: [Asterisk-Users] Zaptel/PRI problem
Tan, My warnings & notices have stopped completely. I was getting them once every few hours though, instead of once every few minutes. However, in the process of trying to resolve issues, we also rolled back the codebase to 3/5/2004. This *maybe* a red herring, but I see several other people on the list experiencing problems which went away after a 3/5/2004 roll-back. Now, ours did not (initially) go away. But the rollback + interrupts seems to at-least stop the spurious PRI errors. WW - Original Message Follows - > Here is what we get: > > Apr 10 18:10:34 WARNING[-1179604048]: chan_zap.c:6026 > zt_pri_error: PRI: Read on 24 failed: Unknown error 500 > Apr 10 18:10:34 NOTICE[-1179604048]: chan_zap.c:6740 > pri_dchannel: PRI got event: 8 on span 1 > > We were getting around 5 messages per second. I turned off > the usb interface in the bios and now the messages have > greatly reduced in number e.g. a few messages each minute. > > We only see the errors on the console. What frequency of > messages were you getting? Sound dropping in voice calls > may be related to something else. > > Tan > > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf > Of [EMAIL PROTECTED] > Sent: 10 April 2004 17:51 > To: [EMAIL PROTECTED] > Subject: RE: [Asterisk-Users] Zaptel/PRI problem > > > Tan, > Scary ... > What we used to see was quite few (and sporadic) notices > and warnings in the /var/log/messages file reporting PRI > trouble. Especially event 6 and event 8 if I recall. > These notices and warnings have disappeared since we > resolved the interrupt issue. Because the implementation > is an office PBX, and the office had light call volume > Friday afternoon and closed for the weekend, I do NOT know > if users might still experience sound dropping on voice > calls. My question to you: when you state 'similar > problems', you mean dropping voice (sound), or having > spurious PRI events, or both. > Cheers, > WW > > - Original Message Follows - > > Hi, > > > > Glad that your problem was solved, but we are still > > exerpiencing a similar problem but our interrupts show: > > > > 0: 162034 XT-PIC timer > > 1:234 XT-PIC keyboard > > 2: 0 XT-PIC cascade > > 5:782 XT-PIC eth1 > > 7: 7448 XT-PIC eth0 > > 8: 1 XT-PIC rtc > > 10: 0 XT-PIC usb-ohci > > 11: 19330 XT-PIC ide0, ide1 > > 12:468 XT-PIC PS/2 Mouse > > 14: 0 XT-PIC ide2 > > 15: 967550 XT-PIC t1xxp > > NMI: 1 > > ERR: 0 > > > > Any ideas? > > Tan > > > > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf > > Of [EMAIL PROTECTED] > > Sent: 10 April 2004 17:23 > > To: [EMAIL PROTECTED] > > Subject: Re: [Asterisk-Users] Zaptel/PRI problem > > > > > > To the list ... > > The problem appears to be fixed. Short answer: > > interrupts. When looking at cat /proc/interrupts it was > > seen that the wct1xxp card was sharing interrupts with > > a.o. the eth0 main ethernet driver. The solution we > > simple: move the T1 card to another PCI slot. Upon > > checking cat /proc/interrupts again, the card now had > > its very own interrupt. So far I have NOT seen any PRI > > notices or warnings. User experience has been limited > due to a light workload on Friday, and, of > > > course, this being Easter weekend. Since this is such a > > recurring problem (from witnessing list postings), > > perhaps digium should have a notice or something about > > this when they ship the cards out. Especially a > > relative newbie (first T1 on an asterisk box) like me is > > likely to even look at this interrupts situation. > > Especially when there are other issues (zaptel.conf and > > zapata.conf) to get a functional system. > > Maybe this deserves a wiki input. In any case, > > Happy Easter > > WW > > > > - Original Message Follows - > > > Dimitri, > > > I just got off the phone with digium. Here's what I > > > (from my notes) the event codes mean > > > Event 4: Alarm detected > > > Event 5: Alarm cleared > > > Event 6: Abort HDLC Frams > > > Event 8: Bad HCS > > > The 6 & 8 which occur sporadically are possibly > > > causing the observed symptoms. > > >
RE: [Asterisk-Users] Zaptel/PRI problem
Tan, Scary ... What we used to see was quite few (and sporadic) notices and warnings in the /var/log/messages file reporting PRI trouble. Especially event 6 and event 8 if I recall. These notices and warnings have disappeared since we resolved the interrupt issue. Because the implementation is an office PBX, and the office had light call volume Friday afternoon and closed for the weekend, I do NOT know if users might still experience sound dropping on voice calls. My question to you: when you state 'similar problems', you mean dropping voice (sound), or having spurious PRI events, or both. Cheers, WW - Original Message Follows - > Hi, > > Glad that your problem was solved, but we are still > exerpiencing a similar problem but our interrupts show: > > 0: 162034 XT-PIC timer > 1:234 XT-PIC keyboard > 2: 0 XT-PIC cascade > 5:782 XT-PIC eth1 > 7: 7448 XT-PIC eth0 > 8: 1 XT-PIC rtc > 10: 0 XT-PIC usb-ohci > 11: 19330 XT-PIC ide0, ide1 > 12:468 XT-PIC PS/2 Mouse > 14: 0 XT-PIC ide2 > 15: 967550 XT-PIC t1xxp > NMI: 1 > ERR: 0 > > Any ideas? > Tan > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf > Of [EMAIL PROTECTED] > Sent: 10 April 2004 17:23 > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Zaptel/PRI problem > > > To the list ... > The problem appears to be fixed. Short answer: > interrupts. When looking at cat /proc/interrupts it was > seen that the wct1xxp card was sharing interrupts with > a.o. the eth0 main ethernet driver. The solution we > simple: move the T1 card to another PCI slot. Upon > checking cat /proc/interrupts again, the card now had its > very own interrupt. So far I have NOT seen any PRI > notices or warnings. User experience has been limited due > to a light workload on Friday, and, of course, this being > Easter weekend. Since this is such a recurring problem > (from witnessing list postings), perhaps digium should > have a notice or something about this when they ship the > cards out. Especially a relative newbie (first T1 on an > asterisk box) like me is likely to even look at this > interrupts situation. > Especially when there are other issues (zaptel.conf and > zapata.conf) to get a functional system. > Maybe this deserves a wiki input. In any case, > Happy Easter > WW > > - Original Message Follows - > > Dimitri, > > I just got off the phone with digium. Here's what I > > (from my notes) the event codes mean > > Event 4: Alarm detected > > Event 5: Alarm cleared > > Event 6: Abort HDLC Frams > > Event 8: Bad HCS > > The 6 & 8 which occur sporadically are possibly causing > > the observed symptoms. > > Now ... what causes 6 & 8 is the question. > > Interrupt conflicts was one suggested possibility. > > Another possibility is 'stuff' from the Telco which is > > not understood / mis-understood by the driver. > > I'll keep the list posted. > > Willy > > > > - Original Message Follows - > > > Dear Willy > > > i notice the same problem with my E100P using the > > > latest cvs zaptel driver i have try every type of > > > config in /etc/zaptel.conf to check if i have missed > > > something in timing conf but nothing... Digium > > > help... :-) thanks in advance Dimitri > > > > > > On Thursday 08 April 2004 23:07, [EMAIL PROTECTED] > > > > wrote: Chris, > > > > Thank you for posting this. Since it concerns my > > > > 'production' system, let me comment. After > > > > 'downshifting' to a previous release (for no good > > > > reason other than desperation and teh fact that an > > > > earlier list entry had commented that it cleared up > > > > the problems) I am sad to report that the system > > > > failed again. Miscellaneous throughout the day: Apr > > > > 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 > > > > failed: Unknown error 500 Apr 8 13:41:27 > > > > NOTICE[-1210639440]: PRI got event: 8 on span 1 > > > > Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on > > > > 32 failed: Unknown error 500 > > > > Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: > > > > 6 on span 1 > > > > Apr 8 13:42:07 WARNING[-1210639440]: PRI: Read on > > > > 32 fail
Re: [Asterisk-Users] Zaptel/PRI problem
To the list ... The problem appears to be fixed. Short answer: interrupts. When looking at cat /proc/interrupts it was seen that the wct1xxp card was sharing interrupts with a.o. the eth0 main ethernet driver. The solution we simple: move the T1 card to another PCI slot. Upon checking cat /proc/interrupts again, the card now had its very own interrupt. So far I have NOT seen any PRI notices or warnings. User experience has been limited due to a light workload on Friday, and, of course, this being Easter weekend. Since this is such a recurring problem (from witnessing list postings), perhaps digium should have a notice or something about this when they ship the cards out. Especially a relative newbie (first T1 on an asterisk box) like me is likely to even look at this interrupts situation. Especially when there are other issues (zaptel.conf and zapata.conf) to get a functional system. Maybe this deserves a wiki input. In any case, Happy Easter WW - Original Message Follows - > Dimitri, > I just got off the phone with digium. Here's what I (from > my notes) the event codes mean > Event 4: Alarm detected > Event 5: Alarm cleared > Event 6: Abort HDLC Frams > Event 8: Bad HCS > The 6 & 8 which occur sporadically are possibly causing > the observed symptoms. > Now ... what causes 6 & 8 is the question. > Interrupt conflicts was one suggested possibility. > Another possibility is 'stuff' from the Telco which is not > understood / mis-understood by the driver. > I'll keep the list posted. > Willy > > - Original Message Follows - > > Dear Willy > > i notice the same problem with my E100P using the > > latest cvs zaptel driver i have try every type of > > config in /etc/zaptel.conf to check if i have missed > > something in timing conf but nothing... Digium help... > > :-) thanks in advance Dimitri > > > > On Thursday 08 April 2004 23:07, [EMAIL PROTECTED] > > > wrote: Chris, > > > Thank you for posting this. Since it concerns my > > > 'production' system, let me comment. After > > > 'downshifting' to a previous release (for no good > > > reason other than desperation and teh fact that an > > > earlier list entry had commented that it cleared up > > > the problems) I am sad to report that the system > > > failed again. Miscellaneous throughout the day: > > > Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 > > > failed: Unknown error 500 > > > Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: 8 > > > on span 1 > > > Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 > > > failed: Unknown error 500 > > > Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: 6 > > > on span 1 > > > Apr 8 13:42:07 WARNING[-1210639440]: PRI: Read on 32 > > > failed: Unknown error 500 > > > Apr 8 13:42:07 NOTICE[-1210639440]: PRI got event: 8 > > > on span 1 > > > Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read on 34 > > > failed: Unknown error 500 > > > Apr 8 16:44:01 NOTICE[-1210631248]: PRI got event: 6 > > > on span 1 > > > Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read on 34 > > > failed: Unknown error 500 > > > Apr 8 16:44:01 NOTICE[-1210631248]: PRI got event: 6 > > > on span 1 > > > > > > Then this -- possibly not related ? > > > > > > Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries > > > exceeded on call [EMAIL PROTECTED] for > > > seqno 102 (Request) > > > Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries > > > exceeded on call [EMAIL PROTECTED] for > > > seqno 103 (Request) > > > Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries > > > exceeded on call [EMAIL PROTECTED] for > > > seqno 104 (Request) > > > Apr 8 16:51:46 WARNING[-1137157200]: Maximum retries > > > exceeded on call [EMAIL PROTECTED] for > > > seqno 105 (Request) > > > > > > And finally, I'll show you a RESTART log > > > > > > Apr 8 17:41:32 WARNING[-1085030272]: Ignoring port > > > for now Apr 8 17:41:33 WARNING[-1085030272]: XXX I > > > don't work right with non-full duplex sound cards XXX > > > Apr 8 17:41:33 WARNING[-1189983312]: Read error on > > > sound device: Resource temporarily unavailable > > > Apr 8 17:41:33 ERROR[-1085030272]: Unable to load > > > config iax1.conf > > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > > channel 1 > > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > > channel
Re: [Asterisk-Users] Zaptel/PRI problem
Dimitri, I just got off the phone with digium. Here's what I (from my notes) the event codes mean Event 4: Alarm detected Event 5: Alarm cleared Event 6: Abort HDLC Frams Event 8: Bad HCS The 6 & 8 which occur sporadically are possibly causing the observed symptoms. Now ... what causes 6 & 8 is the question. Interrupt conflicts was one suggested possibility. Another possibility is 'stuff' from the Telco which is not understood / mis-understood by the driver. I'll keep the list posted. Willy - Original Message Follows - > Dear Willy > i notice the same problem with my E100P using the > latest cvs zaptel driver i have try every type of config > in /etc/zaptel.conf to check if i have missed something > in timing conf but nothing... Digium help... :-) thanks in > advance Dimitri > > On Thursday 08 April 2004 23:07, [EMAIL PROTECTED] wrote: > > Chris, > > Thank you for posting this. Since it concerns my > > 'production' system, let me comment. After > > 'downshifting' to a previous release (for no good reason > > other than desperation and teh fact that an earlier list > > entry had commented that it cleared up the problems) I > > am sad to report that the system failed again. > > Miscellaneous throughout the day: > > Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 > > failed: Unknown error 500 > > Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: 8 on > > span 1 > > Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 > > failed: Unknown error 500 > > Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: 6 on > > span 1 > > Apr 8 13:42:07 WARNING[-1210639440]: PRI: Read on 32 > > failed: Unknown error 500 > > Apr 8 13:42:07 NOTICE[-1210639440]: PRI got event: 8 on > > span 1 > > Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read on 34 > > failed: Unknown error 500 > > Apr 8 16:44:01 NOTICE[-1210631248]: PRI got event: 6 on > > span 1 > > Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read on 34 > > failed: Unknown error 500 > > Apr 8 16:44:01 NOTICE[-1210631248]: PRI got event: 6 on > > span 1 > > > > Then this -- possibly not related ? > > > > Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries > > exceeded on call [EMAIL PROTECTED] for > > seqno 102 (Request) > > Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries > > exceeded on call [EMAIL PROTECTED] for > > seqno 103 (Request) > > Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries > > exceeded on call [EMAIL PROTECTED] for > > seqno 104 (Request) > > Apr 8 16:51:46 WARNING[-1137157200]: Maximum retries > > exceeded on call [EMAIL PROTECTED] for > > seqno 105 (Request) > > > > And finally, I'll show you a RESTART log > > > > Apr 8 17:41:32 WARNING[-1085030272]: Ignoring port for > > now Apr 8 17:41:33 WARNING[-1085030272]: XXX I don't > > work right with non-full duplex sound cards XXX > > Apr 8 17:41:33 WARNING[-1189983312]: Read error on > > sound device: Resource temporarily unavailable > > Apr 8 17:41:33 ERROR[-1085030272]: Unable to load > > config iax1.conf > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 1 > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 2 > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 3 > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 4 > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 5 > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 6 > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 7 > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 8 > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 9 > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 10 > > Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on > > channel 11 > > Apr 8 17:41:38 WARNING[-1210963024]: PRI: Read on 32 > > failed: Unknown error 500 > > Apr 8 17:41:38 NOTICE[-1210963024]: PRI got event: 5 on > > span 1 > > Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on > > channel 1: Red Alarm > > Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on > > channel 2: Red Alarm > > Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on > > channel 3: Red Alarm > > Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on > > channel 4: Red Alarm > > Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on > > channel 5: Red Alarm
Re: [Asterisk-Users] Zaptel/PRI problem
Chris, Thank you for posting this. Since it concerns my 'production' system, let me comment. After 'downshifting' to a previous release (for no good reason other than desperation and teh fact that an earlier list entry had commented that it cleared up the problems) I am sad to report that the system failed again. Miscellaneous throughout the day: Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 failed: Unknown error 500 Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: 8 on span 1 Apr 8 13:41:27 WARNING[-1210639440]: PRI: Read on 32 failed: Unknown error 500 Apr 8 13:41:27 NOTICE[-1210639440]: PRI got event: 6 on span 1 Apr 8 13:42:07 WARNING[-1210639440]: PRI: Read on 32 failed: Unknown error 500 Apr 8 13:42:07 NOTICE[-1210639440]: PRI got event: 8 on span 1 Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read on 34 failed: Unknown error 500 Apr 8 16:44:01 NOTICE[-1210631248]: PRI got event: 6 on span 1 Apr 8 16:44:01 WARNING[-1210631248]: PRI: Read on 34 failed: Unknown error 500 Apr 8 16:44:01 NOTICE[-1210631248]: PRI got event: 6 on span 1 Then this -- possibly not related ? Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Request) Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 103 (Request) Apr 8 16:51:45 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 104 (Request) Apr 8 16:51:46 WARNING[-1137157200]: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 105 (Request) And finally, I'll show you a RESTART log Apr 8 17:41:32 WARNING[-1085030272]: Ignoring port for now Apr 8 17:41:33 WARNING[-1085030272]: XXX I don't work right with non-full duplex sound cards XXX Apr 8 17:41:33 WARNING[-1189983312]: Read error on sound device: Resource temporarily unavailable Apr 8 17:41:33 ERROR[-1085030272]: Unable to load config iax1.conf Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 1 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 2 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 3 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 4 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 5 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 6 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 7 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 8 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 9 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 10 Apr 8 17:41:38 NOTICE[-1221452880]: Alarm cleared on channel 11 Apr 8 17:41:38 WARNING[-1210963024]: PRI: Read on 32 failed: Unknown error 500 Apr 8 17:41:38 NOTICE[-1210963024]: PRI got event: 5 on span 1 Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 1: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 2: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 3: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 4: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 5: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 6: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 7: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 8: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 9: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 10: Red Alarm Apr 8 17:49:10 WARNING[-1221452880]: Detected alarm on channel 11: Red Alarm Apr 8 17:49:10 WARNING[-1210963024]: PRI: Read on 32 failed: Unknown error 500 Apr 8 17:49:10 NOTICE[-1210963024]: PRI got event: 4 on span 1 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 1 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 2 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 3 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 4 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 5 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 6 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 7 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 8 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 9 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 10 Apr 8 17:49:18 NOTICE[-1221452880]: Alarm cleared on channel 11 Apr 8 17:49:18 WARNING[-1210963024]: PRI: Read on 32 failed: Unknown error 500 Apr 8 17:49:18 NOTICE[-1210963024]: PRI got event: 5 on span 1 Can someone PLEASE interpret these messages? BTW: Calls & messages to digium support have gone unanswered :( TIA Willy - Original Message Follows - > Just a note to the list, this problem still seems to exist > as late as 2004-04-07. > > Symptoms as we have seen it are this. System runs along > just fine
Re: [Asterisk-Users] Stable Relase Broken ?
OOPS ... > > Just add a ",r" option to the Dial statement, or, do I get it ;) Thnx, WW Willy Wouters ypOne Publishing ___ 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] Stable Relase Broken ?
Nah you missed my point !>> It is the person CALLING IN who does not hear the ringing. Stepping back to a previous downloaded CVS release fixed that problem. Since I have a system actualy being used by a salesforce who gets reeeaaally uptight when their phones don't work, I guess I'll have to wait until the 'stable' release is realy 'stable' ;) WW - Original Message Follows - > > All, > > I upgraded to the [*] stable release branch. > > When I call into the box (confirmed on 2 installations) > > the caller no longer hears the ringing. The CLI > > confirms that extensions are being 'rung'. > > Whassup? > > Just add a ",r" option to the Dial statement, or, do > another upgrade. > > > > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Stable Relase Broken ?
More Info: And I went back to CVS-03/26/04 and can hear the 'ringing' again when I call in to the box ... BTW: This behavior exists on the production system (T1 PRI interface to PSTN only) and on the Developent system (FXO/FXS and IAX2 interfaces) Cheers, Willy - Original Message Follows - > All, > I upgraded to the [*] stable release branch. > When I call into the box (confirmed on 2 installations) > the caller no longer hears the ringing. The CLI confirms > that extensions are being 'rung'. > Whassup? > Willy > > Willy Wouters > ypOne Publishing > > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Stable Relase Broken ?
All, I upgraded to the [*] stable release branch. When I call into the box (confirmed on 2 installations) the caller no longer hears the ringing. The CLI confirms that extensions are being 'rung'. Whassup? Willy Willy Wouters ypOne Publishing ___ 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 connect to voicemail
The snom dials into an account caled 'asterisk'
Exten => asterisk,1,Answer,1
Exten => asterisk,2,Wait,1
Exten => asterisk,3,Voicemailmain(${CALLERIDNUM})
- Original Message Follows -
> I think this is what you are looking for
>
> Exten => 1000,1,Answer,1
> Exten => 1000,2,Wait,1
> Exten => 1000,3,Voicemailmain([EMAIL PROTECTED])
> - Original Message -
> From: Mitchell S. Sharp
> To: [EMAIL PROTECTED]
> Sent: Monday, April 05, 2004 5:27 PM
> Subject: Re: [Asterisk-Users] Auto connect to voicemail
>
>
> On Mon, 2004-04-05 at 15:57, Brian Rathman wrote:
> I have the voicemail setup working in that I get the MWI
> and it emails the message correctly. When I pressed the
> MWI button on my SNOM 200, it dials into the voicemail
> system and prompts me for a mailbox and password. I know
> there is a way to automatically connect directly into the
> mailbox via the extension.conf file, but I can not find
> the documentation I am looking for in reference to
> variables and macros for the extensions file. Can someone
> please help me with this issue?
>
> Thanks,
> Brian
> Brian,
>
> At the CLI, type 'show application VoiceMailMain'. You
> can use the CLI 'show applications' command to list all
> available apps. If you hit tab, it acts just like BASH's
> auto complete. Wonderful feature!
>
> Mitch Sharp
> Innovative Solutions
Willy Wouters
ypOne Publishing
___
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[Asterisk-Users] Voicemail Name recording etc
Hi all .. Maybe I am just missing something, but when I press '0' and then '3' to record my name, it gives me an 'after the tone .. '. Then I say my name and press #. It says: your message has been saved. So HOW do I listen to my recording, make sure it sounds OK, and then CONFIRM that is what I want the mailsystem to use OR CANCEL out and leave things the way they are? The same goes, of course, for the unavailable and busy greetings TIA Willy Willy Wouters ypOne Publishing ___ 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 entered digits ignored during wait....
That's what the 'silence' files were invented for. See loligo.com (forgot the exact reference, but do a wiki for J Todd's sound files). Yes, it's a hack, but it works. Cheers, Willy - Original Message Follows - > Greetings, > > > > Below is part of the contents of my extensions.conf file. > > > > exten => s,1,Wait,1 ; Wait a > second before answering. > > exten => s,2,Answer > > exten => s,3,ResponseTimeout,10 ; Set > the amount of time the user > > > ; has to make a selection. > > exten => s,4,DigitTimeout,5; Set the > amount of time user has > > > ; between each number entry when > > > ; dialing an extension. > > exten => s,5,Background(welcome) > > exten => s,6,Background(parties) > > exten => s,7,Wait(10) > > exten => s,8,Background(parties) > > exten => s,9,Wait(10) > > exten => s,10,Background(vm-goodbye) > > exten => s,11,Hangup > > > > I can make a menu selection as long as Background is > running however during Wait(10) DTMF digits are ignored. > How can I wait for a response and register the response at > the same time? I supposed I could create a sound file of > 10 second duration and play this but that seems kinda like > a hack to me. > > > > Gene Kochanowky > > Willy Wouters ypOne Publishing ___ 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] Multiple IAX "register" lines?
Works like a charm for me. I have both VoicePulse and NuPhone registered in IAX. Depending upon the phone nr dialed, I send a call via NP or VP. And yes, my [*] box is behind a NAT. Include the relevant lines of your iax.conf so we can take a look. Cheers, Willy - Original Message Follows - > Hi, > > Having a small problem here and wondering if anyone else > has seen it.. > > My Asterisk box is behind NAT so I need to "register" with > the external IAX Asterisk boxes for calls to be > received.. > > Up till yesterday I only needed to "register" with a > single external IAX server and all was working fine.. Now > I need to "register" with a second external IAX server.. > > So I now have two "register" lines in my IAX.conf.. The > problem is that Asterisk only uses the first one and is > ignoring the second.. If I comment out the first one then > asterisk will happily use the second one but it does not > seem to be happy using both at the same time.. > > Is anyone else having this problem? is it a bug? > > Thanks.. > > ___ > 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 Willy Wouters ypOne Publishing ___ 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
Well ... For starters, in your sip.conf you have dtmfmode=rfc2833 but your phone setup gives send_dtmf=in-audio In your post (below) you also left out authenticate_password=gol but that may be an oversight? BTW: My GS setup uses dtmfmode=info (in my sip.conf for each phone) and send_dtmf=SIP_IPNFO in the phone config Cheers, Willy - Original Message Follows - > Hi there, > I am still trying to make the asterisk SIP proxy server > work with my Grandstream 100 IP phones. > I tried Stephen advice and it did not work. I stil got the > 404 error message. So, rigth now, I am trying the > following configuration(sip.conf): > > ### > ; > ; SIP Configuration for Asterisk > ; > [general] > port = 5060 ; Port to bind to > bindaddr = 0.0.0.0 ; Address to bind to > ;externip = 200.201.202.203 ; Address that we're going to > put in SIP messages if we're behind a NAT > ;localnet = 192.168.0.0 ; Internal NETWORK address > ;localmask = 255.255.255.0 ; Internal netmask > context = default ; Default for incoming calls > ;srvlookup = yes ; Enable SRV lookups on outbound calls > ;pedantic = yes ; Enable slow, pedantic checking for > Pingtel ;tos=lowdelay > ;tos=184 > ;maxexpirey=3600 ; Max length of incoming registration we > allow ;defaultexpirey=120 ; Default length of > incoming/outoing registration ;notifymimetype=text/plain ; > Allow overriding of mime type in NOTIFY ;videosupport=yes > ; Turn on support for SIP video ;disallow=all ; Disallow > all codecs ;allow=ulaw ; Allow codecs in order of > preference dtmfmode=rfc2833 > disallow=all > allow=ulaw > allow=alaw > ;allow=ilbc > > ;register => [EMAIL PROTECTED] ; Register with a SIP > provider ;register => [EMAIL PROTECTED]/1234 ; > Register 2345 at sip provider as 1234 here. > ; > ;[snomsip] > ;type=friend > ;secret=blah > ;host=dynamic > ;dtmfmode=inband ; Choices are inband, rfc2833, or info > ;defaultip=192.168.0.59 > ;mailbox=1234,2345 ; Mailbox for message waiting > indicator ;restrictcid=yes ; To have the callerid > restriced -> sent as ANI > > ;[pingtel] > ;type=friend > ;username=pingtel > ;secret=blah > ;host=dynamic > ;qualify=1000 ; Consider it down if it's 1 second to > reply ;callgroup=1,3-4 > ;pickupgroup=1,3-4 > ;defaultip=192.168.0.60 > > ;[cisco] > ;type=friend > ;username=cisco > ;secret=blah > ;nat=yes ; This phone may be natted > ;host=dynamic > ;canreinvite=no ; Cisco poops on reinvite sometimes > ;qualify=200 ; Qualify peer is no more than 200ms away > ;defaultip=192.168.0.4 > > ;[cisco1] > ;type=friend > ;username=cisco1 > ;fromuser=markster ; Specify user to put in "from" > instead of callerid ;secret=blah > ;host=dynamic > ;defaultip=192.168.0.4 > ;amaflags=default ; Choices are default, omit, billing, > documentation ;accountcode=markster ; Users may be > associated with an accountcode tp ease billing > > > [1001] > type = friend > context = default > secret = gol > host = dynamic > callerid = "STREAM-1001" <1001> > ;dtfmmode=inband > canreinvite=no > defaultip=192.168.0.105 > > > [1002] > type = friend > context = default > secret = gol > host = dynamic > callerid = "STREAM-1002" <1002> > ;dtfmmode=inband > canreinvite=no > defaultip=192.168.0.104 > ## > > This is the configuration of my SIP-phones: > > > ipaddr=192.168.0.105 > sipserver=192.168.0.102 > sipserver_port=5060 > outboundproxy=null > outboundproxy_port=null > userid=1001 > authenticateid=1001 > 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 > send_dtmf=in-audio > dtmf_payload_type=101 > time_zone=GMT-0 > > ipaddr=192.168.0.104 > sipserver=192.168.0.102 > sipserver_port=5060 > outboundproxy=null > outboundproxy_port=null > userid=1004 > authenticateid=1004 > 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 > send_dtmf=in-audio > dtmf_payload_type=101 > time_zone=GMT-0 > > > What's wrong here?? > > When I try to dial from one phone to the other, I get 404 > error message. > > Please, somebody help me. > > > ___ > 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 Willy Wouters ypOne Publishing ___ 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 fails to detect user hung up
Ron, It is a multi-reported problem, yet no resolution. I would suggest it is a bug. I have had intermittent success with POTS provided by AllTel in Texas. My opinion, you're SOL and there is very little you can do. I keep hoping that someone at digium will pick up on this and look at the hardware design etc. BTW, I tried kewlstart, loopstart etc. and it doesn't make any difference. As I said, it's intermittent on POTS, and it's constant on my ISDN fxs channels. Cheers, Willy - Original Message Follows - > > I am using the wildcard X100P with *. PSTN line comes in > to the FXO port of this card. Everything works fine most > of the time. However, occasionally Asterisk doesn't seem > to be able to detect the user has hung up and therefore > tie up the line for quite a long time. Does anyone know if > there's anything I can do to fix this problem? > > thanks > > Ron > Willy Wouters ypOne Publishing ___ 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] Phones can talk to asterisk but not each other through it
Tony, What is the BW connectivity at the [*] box? You may try to set the GS phones to GSM codec to reduce BW, and see if that improves the situation. WW - Original Message Follows - > I posted this a week or two ago but no replies, so trying > again... > > Summary: Two phones in different locations, each behind > NAT, can both talk to an Asterisk server on the net, for > the demo or for voicemail, but can't maintain a call to > each other via that asterisk. > > Original post with details: > > I have a problem with an installation of asterisk on my > colo server. I have a Grandstream BT102 behind a Linux NAT > firewall, and my colleague also has one behind his. > > My connection is ADSL with 512k down and 256k up. My > colleague's is Cable with 600k down and I don't know > whether it's 128k or 256k up. > > I have the phones set up in sip.conf with nat=yes, > qualify=yes and canreinvite=no. Each phone can > successfully connect with Asterisk and dial the Asterisk > Demo, leave and pick up voicemail, etc. > > However, if one phone tries to dial the other, once the > called phone is answered, the audio starts off very > stuttery and broken, and after a few seconds dies > completely and the call gets dropped. > > In the asterisk log there are many entries for that time > saying: Recv error: Resource temporarily unavailable. > > I am using the zaprtc timer module on the asterisk server, > but in any case I understood that was only required for > MeetMe or MOH. > > The server system is a Duron XP 1800, with 512MB RAM, > running Fedora Core 1 with updates, and a standard 2.4.22 > kernel that was recompiled only to make the RTC a module > instead of compiled in (so I could rmmod it and then load > zaprtc instead, which works fine). > > Can anyone suggest what things I should check or change? > > Cheers > 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 Willy Wouters ypOne Publishing ___ 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] Information Needed
Check the wiki http://voip-info.org WW - Original Message Follows - > Hello there,I am new to Asterisk. This is my first day on > it. Can someone tell me minimum hardware requirements > (computerwise) and which version on Linux i need to run > it. Thanks. > > ___ > Join Excite! - http://www.excite.com > The most personalized portal on the Web! > Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 200 Voice Call / Paging
Christian, I guess I am Confused about the 'header' stuff. I am using the SIP strictly on a LAN as extensions to the [*]. Hence, I have a line in sip.conf like this: [2200] ;snom 200 callerid=Reception <2200> type = friend disallow=all allow=ulaw allow=alaw username = 2200 secret = 2200 host = dynamic dtmfmode = rfc2833 context=intern mailbox = 2200 In extensions.conf I have exten => 2200,1,Dial(SIP/2200,20,tT) Now, [*] is at 192.168.1.16. Where does the 'header' you refer to get sent? I tried adding intercom=true to the sip.conf but that is not it right? Lost ... Willy - Original Message Follows - > To use "Intercom" mode in the current releases of the snom > 200, you need to use an "intercom=true" flag in the > To-Header. Essentially that makes the phone to pick up the > call immediately. > > To: > > However, this mechanism is likely to change because of > security concerns and new interoperable methods. > > Christian > > > -Original Message- > > From: [EMAIL PROTECTED] > > [mailto:asterisk-users- [EMAIL PROTECTED] On > > Behalf Of [EMAIL PROTECTED] Sent: Sunday, March 21, > > 2004 5:25 PM To: [EMAIL PROTECTED] > > Subject: [Asterisk-Users] Snom 200 Voice Call / Paging > > > > To All, > > Several months (2003) ago there was a discussion > > regarding overhead paging & intercom functionality with > > SIP / Asterisk. Jerry Gibson, John Todd and various > > others participated (from checking the archives). One > > person even responded that they had the stuff working > > with the snom 200s. > > Voice Call (i.e. on-hook speaker/mic) is realy important > > in a lot of apps. It would appear that the snom 200 and > > by extension the snom 105 support the functionality. > > I will be happy to make a wiki entry to explain & demo > > this functionality once I have it working properly. I > > also understand that the (mis)use of conferencing is > > frowned upon as it wastes bandwidth and CPU. However, > > until a better way comes around, that is not a problem > > as there are quite a few applications where (a) one > > needs Voice Call (which is 1 <-> 1) and / or an > > 'allPage' which can be limited to a subset of all > > phones. Typically phones which are in designated or > > public areas, conference rooms, etc. The BW/CPU issue > can be controlled. Better a limited solution than no > > solution at all ;) > > I am also allowing for the limitation that all > > participating phones are on the same LAN with the [*]. > > Anyone who has this successfully working with snom, > > please respond .. Using the [*] sound card for a > > separate PA system is NOT an option ;) > > As I said, I will be 'distilling' the info and turn it > > into a wiki entry. > > Cheers and TIA, > > Willy > > > > Willy Wouters > > ypOne Publishing > > > > ___ > > 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 Willy Wouters ypOne Publishing ___ 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 Tone-based Supervisory Disconnect ?
The X100P hangup problem is indeed pervasive. My current testbed has the X100P connecting to an FXS breakout of a dual ISDN channel box. Indeed, remote hangup is NOT detected. When I switched it to a POTS line, all the sudden it seemed to work OK. This is a serious limitation in some scenarios however. There was a response a few days ago about recompiling [*] with some different options in the driver (had to do with the same problem on a UK line). When this gets resolved, this would make for another fine wiki addition. Cheers, WW - Original Message Follows - > Hello all, > Through the previous two weeks I have been working on > my first asterisk installation. So far all my doubts and > problems were aswered by the list history or on-line > documentation. I have a two SIP softphones + external > analog line working fine. However, I just came across a > documented problem that as far as I can tell there is no > good answer: the X100P FXOdoes not hung up when the > remote end disconnects. I have a X100P connected to an > extension of my company PBX. When I get an incoming call, > it stays up even when the caller disconnects and I have > to use a soft hungup. > I have already tried all signalling methods (ks, gs, > ls) with no luck. Searching on the web I found a good > description of the problem on the Cisco site: > http://www.cisco.com/warp/public/788/signalling/fxo_disconnect.html > There is a method of hang up detection there called > Tone-based Supervisory Disconnect. It seems to me that it > would fix my problems, because my company PBX does send > some fast tones when the line gets disconnected by the > remote end. > Does asterisk have this feature? How can I turn it on? > > Thanks a lot, > Gelson > > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Important: The Asterisk Mailing list(newsubject)
Strongly Agree :) WW - Original Message Follows - > The only thing I hate more than not having a proper > reply-to on a mailing list (one that replies to the LIST) > is the people who havn't been on the net long enough to > know how mailing lists work, and their whole function. > Mailing lists are communities. The primary function is to > share procedures, patches, fixes, workarounds, programming > knowledge, etc.. with the rest of the community. Its the > rare exception that once in a great while a topic strays > off and goes personal/off-list. This should happen when > the community cannot benefit from the discussion, such as > a private deal for equipment (which is sometimes frowned > upon with lists, but sometimes enjoyed), or some basic > hand-holding that goes beyond the scope of the list, and > that the rest of the listmembers should know. (I.E. > someone asking how to setup Linux, thus not having > anything to do with Asterisk, untill they get to the point > where they can install Asterisk). > > So, my vote is to keep the "reply-to" as going to the > list. Also, don't hijack subjects. If you are going to > use "reply" insted of post, at least re-write the subject > line! > > Please direct all flames privately, where they can be > properly transfered to /dev/null > > - - - Jon Myers > "Online" since 1985 (I know, not longer than alot of > prople, but more than a couple years). > > > > At 07:30 PM 3/21/2004 -0600, you wrote: > >On Fri, 2004-03-19 at 22:23, Darren Nickerson wrote: > >> Folks, > >> > >> I strongly support removing the current reply-to-list > setting, and you >> should too. > >> > >> Like many new list admins, I once thought the reply-to > was kewel. Requests >> to remove it kept coming up, ... > usually around the same time someone >> embarrassed > themselves by posting a personal reply/flame to the list. > >> Someone, in frustration, finally pointed me to the > following URL: >> > >>http://www.unicom.com/pw/reply-to-harmful.html > >> > >> I saw the light. > >> > >> Please can the list admin step in and end this thread > by either: >> > >>a) announcing that the reply-to override has been > removed >>b) announcing their resignation ;-) > > > > > >I'm sorry you saw the wrong light. You are peering into a > light that >will anger many more of us to the point of > removing ourselves from the >list as it becomes impossible > to filter appropriately. > > >Reply to group has a nasty habit of piling up addresses > and then people >who have dropped out of the thread are > still getting barraged by >messages where their address is > still a part of it. > > >It is bad enough we have users too lame to click on a > link to the >submission url and instead just reply and > erase old content, your >suggestion would just make people > more likely to get nailed with >unrelated content. > > > >Open source software thrives by efficient and open > communications. To >start suggesting people take useful > commentary off list by making it >less easy to reply to > the list only reduces our resources. It also >starts a lot > of private communications and possibly private flame wars. > >If you post embarrassing information, or if your post > embarrass you in >public, maybe they didn't need to be > said in the first place. > > >Your aversion to fixing a to line when you take a message > off list is >not worth breaking good mail filtering. > > > >You can probably blame me for the original switch of the > Reply-To >header. I believe I am the one who requested it > soo long ago. >-- > >Steven Critchfield <[EMAIL PROTECTED]> > > > >_______ > >Asterisk-Users mailing list > >[EMAIL PROTECTED] > >http://lists.digium.com/mailman/listinfo/asterisk-users > >To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users Willy Wouters ypOne Publishing ___ 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] If you know your party's extension # please dial it now ...
Please elaborate ...
- Original Message Follows -
> You don't have to avoid using an option 3 when even if
> extensions are 3XXX
>
> > -Original Message-
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf
> > Of Rich Adamson
> > Sent: Sunday, March 21, 2004 4:19 PM
> > To: Asterisk Users
> > Subject: Re: [Asterisk-Users] If you know your party's
> > extension # please dial it now ...
> >
> > > I've built the usual "press one for sales, 2 for
> > > support" IVR which works fine but I'm having
> > > difficulty in allowing callers to type in whole
> > > extension numbers.
> > > My internal extn ranges are 3xxx and 4xxx. I have
> > pasted the IVR below
> > > (just in case someone wants one). The welcome message
> > states callers
> > > should type in the extension number they want or
> > choose from the options.
> > > It seems though that one can only press one number
> > > before the IVR moves to the next step.
> > >
> > > I'm starting to think that if my extn's are 3xxx and
> > 4xxx I can't have
> > > any menu choices beginning with 3 or 4. Would this be
> > correct? If so
> > > how does the received DTMF break out of the IVR and
> > get matched to the
> > > relevant dialplan entry?
> > >
> > >
> > > [mainmenu]
> > > exten => s,1,Answer
> > > exten => s,2,SetMusicOnHold(default)
> > > exten => s,3,DigitTimeout,3
> > > exten => s,4,ResponseTimeout,5
> > > ;SAI menu - 1 for tech support, 2 for voicemail, 3
> > > for echo test exten => s,5,Background(welcomemsg)
> > > exten => s,6,Background(choosemsg)
> > >
> > > ; Sales
> > > exten => 1,1,Dial,SIP/3400|20
> > > exten => 1,2,Voicemail(3400)
> > > exten => 1,3,Goto(mainmenu,s,60
> > >
> >
> > Mark,
> >
> > Here's a partial copy of my ivr, and I too am using the
> > 3xxx extensions.
> > Notice I avoided use of option 3 in the ivr menues.
> >
> > [bus-ivr-main]
> > exten => s,1,Wait,1
> > exten => s,2,Answer
> > exten => s,3,DigitTimeout,5
> > exten => s,4,ResponseTimeout,20
> > exten => s,5,Background(npi-greeting) ; "Thanks for
> > calling press 1 for"
> >
> > exten => 1,1,Goto(local-extns|3014|1) ; Sales exten =>
> > 2,1,Dial(${PHONE1}&${PHONE2},15) ; Technical Services
> > exten => 2,2,Voicemail2(u3000) exten => 2,102
> > ,Voicemail2(b3000) exten => 2,103,Hangup
> > exten => 8,1,Goto(npilist|s|1); Company
> > directory list exten => 9,1,Goto(npitest|s|1);
> > VoIP Testing Menu
> > Rich
> >
> >
> >
> > ___
> > Asterisk-Users mailing list
> > [EMAIL PROTECTED]
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
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>http://lists.digium.com/mailman/listinfo/asterisk-users
Willy Wouters
ypOne Publishing
___
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[Asterisk-Users] Snom 200 Voice Call / Paging
To All, Several months (2003) ago there was a discussion regarding overhead paging & intercom functionality with SIP / Asterisk. Jerry Gibson, John Todd and various others participated (from checking the archives). One person even responded that they had the stuff working with the snom 200s. Voice Call (i.e. on-hook speaker/mic) is realy important in a lot of apps. It would appear that the snom 200 and by extension the snom 105 support the functionality. I will be happy to make a wiki entry to explain & demo this functionality once I have it working properly. I also understand that the (mis)use of conferencing is frowned upon as it wastes bandwidth and CPU. However, until a better way comes around, that is not a problem as there are quite a few applications where (a) one needs Voice Call (which is 1 <-> 1) and / or an 'allPage' which can be limited to a subset of all phones. Typically phones which are in designated or public areas, conference rooms, etc. The BW/CPU issue can be controlled. Better a limited solution than no solution at all ;) I am also allowing for the limitation that all participating phones are on the same LAN with the [*]. Anyone who has this successfully working with snom, please respond .. Using the [*] sound card for a separate PA system is NOT an option ;) As I said, I will be 'distilling' the info and turn it into a wiki entry. Cheers and TIA, Willy Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200
Yeah, as in my reply to yoru earlier message, I don't see '4405' in your sip.conf WW - Original Message Follows - > Here's another funny > * CLI puts put > "-- Registered SIP '4405' at IP.address Port 5060 Expires > 3600 " and within seconds the snomm 200 beeps the MWI goes > on the LCD and the light flashes a call from asterisk "Not > Found" > > Willy if you could let me see you sip and config files, if > you have yours working? I'm very sure it is not a LAN > issue, but a config issue > > thanks in advance > > Barry > > ___ > 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 Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200
Barry, My snom are on the same LAN as asterisk hence ... Now, you can set parameters etc. through the web interface. On the LAN where the snon is/are type in teh IP address in a browser, e.g: http://192.168.1.101 This opens the Web Interface Look in SIP Lines You will get an indication whether the phone (line) is registered. Also, for each line there is a 'Mailbox' entry, which should be the extension to check your mail at. In my case that would be '2999' which rings through to VoiceMailMain. In your case it looks like at least one of the phones thinks its extension is '4405'. That is also the default outgoin gline, i.e. asterisk sees the call as coming from '4405'. Now, unless you get a line (in the snom) setup to respond to 4401 or 4403, I don't see how they could be getting any incoming calls at all. Cheers, WW - Original Message Follows - > From: <[EMAIL PROTECTED]> > > > Please include the sip.conf entry for the phone you have > .. > > SIP Configuration for Asterisk > ; > [general] > port = 5060 > bindaddr = 192.168.0.15 > externip = 24.73.215.62 > localnet = 192.168.0.0 > localmask = 255.255.255.0 > tos = lowdelay > disallow = all > allow = ulaw > allow = all > context = INVALID > > > [4403] > type= friend > username= 4403 > secret = 1234 > nat = yes > host= dynamic > context = toll-access > accountcode = barry > mailbox = 4403 > > > [4401] > type= friend > username= 4401 > secret = 1234 > nat = yes > host= dynamic > context = local-access > accountcode = mark > mailbox = 4401 > > > > Also, from your comments I assume that the snom 200 is > > on the same LAN as the [*] box? > > No they are not on the same LAN > > > On the snom web interface, does it show that line 1 > > (which I assume you are using) is 'registered'? > > Not sure where you see this, First page has Outgoing line: > [EMAIL PROTECTED] > > Sip Line Pages has Name: Phone1 Account: 4405 > Registrar: 24.73.215.62 > Mailbox: 4405Ringer: Ringer2 > > For some reason MWI, wants to dial [EMAIL PROTECTED], > I have not exten or > account "asterisk" ???, can't even find where this is set > ? > > Thanks again > Barry > > > ___ > 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 Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 200
Barry, I also just got a new snom 200. Still discovering features, but as a whole it is working fine. Please include the sip.conf entry for the phone you have .. Also, from your comments I assume that the snom 200 is on the same LAN as the [*] box? On the snom web interface, does it show that line 1 (which I assume you are using) is 'registered'? Willy - Original Message Follows - > Greetings All > > I'm busy trying out my new snom 200(s) > I have it connected and * CLI tells me registered > > 1) I pick up the handset and hear the dial tone > 2) Dial and Ext, that says Date & Time (13) > 3) * CLI scrolls that the call is connected and time is > being spoken >YET the handset is quite and silent? WHY ? > > > Also if I dial for voicemailmain ext (8) > * CLI says connected vm-login > Yet again the handset it silent, ? > > What have I not configured? > sip.conf has > disallow = all > allow = ulaw > > The snom 200 is set to G.711u codec > > What is wrong here, please anyone? > Thanks in advance > > Barry > > > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Speaking of ring tones...
Kev, On my snom 200 the ring volume is controlled with the speaker volume setting. WW - Original Message Follows - > Has anyone figured out how to change the volume of the > ring tone on the snom 200? > > Its pretty loud in our office when a group of these things > start to ring together. > > Kev > > > I kinda like it .. ;) > > Nice & conservative. > > OTOH, the new snom 200 I just got today has some > > reeeaaally weird ring tones (and nothing really > > 'traditional'). Now, maybe we should take a lesson from > > the cell-phone people, and talk manufacturers into > > letting us download ringtone(s). Cheers, WW > > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Speaking of ring tones...
I kinda like it .. ;) Nice & conservative. OTOH, the new snom 200 I just got today has some reeeaaally weird ring tones (and nothing really 'traditional'). Now, maybe we should take a lesson from the cell-phone people, and talk manufacturers into letting us download ringtone(s). Cheers, WW - Original Message Follows - > Anyone know if Grandstream ever plan to implement another > tone on the BT-101? To me, it's very weird hearing > ringback as the ring-in sound. > > Cheers, > 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 Willy Wouters ypOne Publishing ___ 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 Echo was: USB Headsets (Plantronics DSP-400)
Hi, The echo problem is the X100P. The hybrid is 'unbalanced', and basically what happens is that the outgoing sound signal comes right-on back as an incoming signal. The reason you don't notice it using the TDM400P is that the incoming sound is completely 'in-sync' with you talking through the handset. The signal is 'bridged' in the computer, and there is (basically) zero delay. Now, when using the X-Lite, the softphone has to do 'some' processing to get the signal coming in to the wires (X100P) to your computer speakers. This processing introduces delay (latency). Now, you here your voice (outgoing signal, being 'turned-around' at the X100P phoneline interface) coming back at you, but with a more noticable delay. Hence the echo. The same effect also occurs when using SIP hardphones connected to [*] and calling out over the X100P. The solution is to implement a 'hybrid balancing' function at the X100P interface. Traditioinally for FXO this has been done in hardware (not flexible) or more recently in software using embedded DSP chips providing adaptive real-time hybrid balancing. The DSP algorithms are not very complex, and could conceivibly be run on the [*] main processor. I have not looked into the codebase, however. For a system with only a few FXOs it should not be a real problem. Notice, that a T1 interface is digital, and you do not have this 'turn-around' issue. Without looing at the Plantronics design specs, there is no telling whether it would work. In any case, the real answer is to fix the problem right at the X100P interface. Cheers, Willy - Original Message Follows - > Hello all, > > I'm thinking about getting the Plantronics DSP-400 headset > for use with Xlite softphone. I currently have a "analog" > headset that does NOT have a DSP on board, which gives me > mediocre call quality and echo when talking to the PSTN > thru my X100P card. I have zero echo when talking thru my > X100P on my cordless phones attached to the Digium > TDM400P. > > Before I got spend the money I was wondering if others > using USB headsets with a DSP and getting good results? > My thought was thought by using a headset with a DSP on > board the echo would go away? > > Any advice on which USB headset in general to use with a > softphone? > > Thanks in advance, > Ed Rubright > > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Hangup X100P Issues
Yep, I think it's possible a card / driver issue. I tested on POTS (Alltel communications - Texas) and the behavior did not change. Wonder if digium still monitors this list. Cheers, Willy - Original Message Follows - > [EMAIL PROTECTED] wrote: > > > It appears that the X100P (FXO) does somehow not passes > > the 'hangup' signaling *. > > > I am having the same issue on a normal analog POTS line > (but in France so you never know what other signalling > anomalies there may be.) > > The h signal never happens on a POTS dialed in call thru > X100P but works > as expected for SIP. > > r > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Hangup X100P Issues
NOOOP!! Unfortunately, a simple POTS line (AllTel Communications) does not resolve the issue. It appears the problem is somehow related to the digium card, or the drivers or what not. Anyone from digium monitoring this list? Is this a bug thing? FYI here's my zapata.conf - [channels] language=en group=1 signalling=fxs_ks channel => 1 group=2 echocancel = no signalling=fxo_ks mailbox = 2100 callerid = "Fedora" <2100> channel => 2 - and zaptel.conf - loadzone=us defaultzone=us # Load FXS device (a TDM400P) as Channel #, and use kewlstart FXO signalling fxoks=2 # Load FXO device (a T100X) as Channel #, and use kewlstart FXS signalling fxsks=1 - Verry simple setup. Mostly works, but no 'hangup' signaling. Cheers, Willy Original Message Follows - > Ahaa! > I am using a line coming out of an ISDN breakout box .. > I'll try it with a regular analog line next. > I'll let you all know what happens. > Thanks for the hint, > Willy > > - Original Message Follows - > > > > What sort of phone line are you using? Connecting an > > X100P to a PBX line or ISDN TA can cause the problems > > you mention. > > > > Iain > > > > > > --On Wednesday, March 17, 2004 7:37 am -0600 > > [EMAIL PROTECTED] wrote: > > > > > Hullo! > > > It appears that the X100P (FXO) does somehow not > > > passes the 'hangup' signaling *. > > > > > > Sample Scenario 1: > > > I call in on external line X100P. I successfully ring > > > an extension. The extension answers. [we have an > > > established call going on now] I hangup (from the > > > external call). Listening to the extension, I just > > > hear a faitn click and then *silence* as if the caller > > > stopped talking. Eventually the person on the > > > extension will actually hangup, releasing the FXO > > > > > > Sample Scenario 2: > > > As above, I call in through the X100P. I dial an > > > extension for VoiceMailMain. Somewhere in the process > > > , I just hangup. The VoiceMailMain keeps happily > > > looping until *eventually* it actually times out. > > > > > > I have tried both scenarions dialing in through IAX > > > (VoicePulse), and both work as expected: i.e. caller > > > hangs up, callee (on extension) hears a 'click' > > > followed by 'congestion' tone. The 'hangup' event is > > > detected. I searched the archives, but could not find > > > a solution. Any ideas, > > > TIA > > > Willy > > > > > > Willy Wouters > > > ypOne Publishing > > > > > > ___ > > > 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 > > Willy Wouters > ypOne Publishing > > ___ > 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 Willy Wouters ypOne Publishing ___ 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] Hangup X100P Issues
Ahaa! I am using a line coming out of an ISDN breakout box .. I'll try it with a regular analog line next. I'll let you all know what happens. Thanks for the hint, Willy - Original Message Follows - > > What sort of phone line are you using? Connecting an > X100P to a PBX line or ISDN TA can cause the problems you > mention. > > Iain > > > --On Wednesday, March 17, 2004 7:37 am -0600 > [EMAIL PROTECTED] wrote: > > > Hullo! > > It appears that the X100P (FXO) does somehow not passes > > the 'hangup' signaling *. > > > > Sample Scenario 1: > > I call in on external line X100P. I successfully ring > > an extension. The extension answers. [we have an > > established call going on now] I hangup (from the > > external call). Listening to the extension, I just hear > > a faitn click and then *silence* as if the caller > > stopped talking. Eventually the person on the extension > > will actually hangup, releasing the FXO > > > > Sample Scenario 2: > > As above, I call in through the X100P. I dial an > > extension for VoiceMailMain. Somewhere in the process, > > I just hangup. The VoiceMailMain keeps happily looping > > until *eventually* it actually times out. > > > > I have tried both scenarions dialing in through IAX > > (VoicePulse), and both work as expected: i.e. caller > > hangs up, callee (on extension) hears a 'click' followed > > by 'congestion' tone. The 'hangup' event is detected. > > I searched the archives, but could not find a solution. > > Any ideas, > > TIA > > Willy > > > > Willy Wouters > > ypOne Publishing > > > > ___ > > 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 Willy Wouters ypOne Publishing ___ 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] Hangup X100P Issues
Hullo! It appears that the X100P (FXO) does somehow not passes the 'hangup' signaling *. Sample Scenario 1: I call in on external line X100P. I successfully ring an extension. The extension answers. [we have an established call going on now] I hangup (from the external call). Listening to the extension, I just hear a faitn click and then *silence* as if the caller stopped talking. Eventually the person on the extension will actually hangup, releasing the FXO Sample Scenario 2: As above, I call in through the X100P. I dial an extension for VoiceMailMain. Somewhere in the process, I just hangup. The VoiceMailMain keeps happily looping until *eventually* it actually times out. I have tried both scenarions dialing in through IAX (VoicePulse), and both work as expected: i.e. caller hangs up, callee (on extension) hears a 'click' followed by 'congestion' tone. The 'hangup' event is detected. I searched the archives, but could not find a solution. Any ideas, TIA Willy Willy Wouters ypOne Publishing ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI script will not be terminated
To amplify the question: Here's a non-AGI scenario: Call from outside. Dial VoiceMail extension in order to allow the user to check his/her mailbox. User decides this is wrong time, and hangs up. Asterisk keeps looping 4-ever (as witnessed by lookign at console messages) prompting th e(now disconnected) user to select an option. What gives? Cheers, Willy - Original Message Follows - > Hi all, > > if i answer a call on my astbox and go into an AGI > script... then there is somthing happens.(play music or > something like that)...and the person who called to the > box hangs up the script will never be terminated. The > process hangs around self the asterisk process is > terminated. > > Is this a bug, or is there something i'am doing wrong ? > > Thanks for help, > > Thomas. > > ___ > 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] Paging & Intercom
Hi all! Having stuck my neck out and going for [*] for our new sales office, (instead of upgrading a Meridian-Nortel key system), one of the main concerns remains the support (or lack thereof) of paging / intercom functionality. Maybe I am just missing somthing elementary? Here goes: After perusing the list archives, I am more & more concerned about the 'paging' 'voice call' 'intercom' scenario. Quoted below is a message from John Todd (sometime last year) on the list: This is not an explicit answer to all of your questions, but... 1) There is currently no intercom functionality supported by Asterisk as an "in-band" method of communicating with phones. There is the ability to make audio on a phone call appear out of the sound-out port on a soundcard, which may be what you're after if you have a PA system of some sort. 2) You can do everything you're looking for with Asterisk. Spend a bit of money on some hardphones (Cisco ATA-186 is my personal bias, since they have 2 lines and they're cheap) and get an X100P analog adapter. Everything you've mentioned can be demo'ed with that configuraion. JT Let's see if I got this right: (a) single line SIP phones (e.g. GS-BT100) not likely to provide "auto answer" on demand, i.e. allow a 'page' to come through (b) multi line SIP phones: configure an extension for "auto answer". [*] may support all-page by sending voice to all non-busy phones at the same time (c) use ATA-186 - or similar - adapter, and inexpensive ANALOG (2 line?) phone to provide the intercom / pageing functionality For a deployment where there is a relatively large number of phones, the cost-per-phone evidently becomes a principal driver. Here's the user classes: (1) regular user: inexpensive GS BT100 may work. (2) power user / manager (3) shared phones in public areas: i.e. kitchen, computer room (4) attendant Using an old-style paging system is out of the question. Paging must be supported through the hard phones. Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] All-Page in Asterisk
Hi .. When the receptionist parks a call for someone who is in the building but not at their desk, she does an 'allpage' which blares-out over the intercom system: 'Willy you have a call parked at 101'. Willy can then just grab any phone (kitchen, hall, computer room) and pick up the parked call. How do we do an 'allpage' in asterisk? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP - Receptionist
Monastery is neat as a monitoring tool. The console's we're talking about also let the user pick-up calls etc. - Original Message Follows - > See monastery, maybe help you > (http://pbx.unslept.com/newstatus.php) > > Regards, > > Gus > > - Original Message - > From: <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Monday, March 08, 2004 6:27 PM > Subject: [Asterisk-Users] SIP - Receptionist > > > > Hi All! > > I am thinking about fork-lift-upgrading a > > Nortel-Meridian key system with a * PBX driving SIP > > phones in the office. The interface to PSTN would be a > > fractional T1 PRI (11 lines plus D channel). The GS > > phones look acceptable for most users. The forthcoming > > "Sayson 480i" would work for management types. The > > receptionist, however, is currently used seeing a > > backlit display - with buttons - attached to her phone - > > showing all the extensions in the office, and who's has > > a conversation going etc. We believe that autoattendant > > should only be used after hours ;). Question: How do I > > drive - acquire such panels with asterisk? What are they > > called? who makes em? I have seen Monastery, but that > > may be too cumbersome an interface for the relatively > > high call volume. I hope I explained what I am looking > > for. TIA > > WW > > ___ > > 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] SIP - Receptionist
Hi All! I am thinking about fork-lift-upgrading a Nortel-Meridian key system with a * PBX driving SIP phones in the office. The interface to PSTN would be a fractional T1 PRI (11 lines plus D channel). The GS phones look acceptable for most users. The forthcoming "Sayson 480i" would work for management types. The receptionist, however, is currently used seeing a backlit display - with buttons - attached to her phone - showing all the extensions in the office, and who's has a conversation going etc. We believe that autoattendant should only be used after hours ;). Question: How do I drive - acquire such panels with asterisk? What are they called? who makes em? I have seen Monastery, but that may be too cumbersome an interface for the relatively high call volume. I hope I explained what I am looking for. TIA WW ___ 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] message lights and stutter tones
Simon, Do the GS phones support stutter tone as-well-as the message light? I am thinking about buying a load of GS-102's for the office. Any other comments appreciated. TIA Willy - Original Message Follows - > Haha > > The magic tweak,, I knew there had to be one. > That works great thanks > > Simon > > > > Simon Chappell wrote: > >> Hi al > >> > >> I have 3 GS 101's plugged into asterisk. > >> They work great and teh quality of sound I can not > fault. Most people I >> am > >> speaking to now ask if I have a new phone because the > quality is so much >> better. > > > > Don't ever use a Cisco phone if you're happy with your > > GS phones right now. ;) > > > >> My latest quandry is to do with the message button and > stuttertones. I >> dont get either.. If i have a message > waiting the only way we know is by >> email.. > >> > >> I have added mailbox=mailboxNumber for each extension > but still no >> avail. > >> > > > > If you're using multiple voicemail contexts, you will > > need to have [EMAIL PROTECTED] > > > > John > > > > ___ > > 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 > > > > --- > Kind Regards > > Simon Chappell > Email : [EMAIL PROTECTED] > WWW : www.isnsuk.com > Phone : 01403268474 > Mobile: 07811409125 > ___ > 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] Re: Help Newbie: TDM Development Kit
Yeah .. Here's an update. It turns out I need to have a zaptel.conf file !?? Ok, so I found that and I now have the followign setup: - >>>> cat zaptel.conf <<<<< loadzone=us defaultzone=us # Load FXS device (a TDM400P) as Channel #, and use kewlstart FXO signalling fxoks=1 # Load FXO device (a T100X) as Channel #, and use kewlstart FXS signalling fxsks=2 - In my zapata.conf file, I have an entry for the TDM400P as follows: [channels] language=en group=1 context=from-sip signalling=fxo_ks mailbox = 2100 callerid = "Fedora" <2100> channel => 1 -- Now, I have two SIP clients on my local LAN, [2000] and [2100]. The mailbox is at [2999] Locally (i.e. on the LAN, everything works, i.e., I can place calls to all my extensions, and leave voice mail. extension [2000] SIP using X-lite on a PC extension [2001] SIP using X-lite on a Laptop (PC) extension [2100] ZAP using the TDM400P board When I run ztcfg -vv I am getting: - Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. - .. which looks OK to me. Now, of course I need to tell asterisk about the FXO card, so I add the following to zapata.conf .. - group=2 signalling=fxs_ks channel => 2 - and when I start asterisk, this is what shows ... Parsing '/etc/asterisk/zapata.conf': Found Registered channel 1, FXO Kewlstart signalling WARNING Unable to specify channel 2: No such device : Unable to open channel 2: No such device here = 0, tmp->channel = 2, channel = 2 msetup_zap: Unable to register channel '2' WARNING: mast_load_resource: load_module failed, returning -1 - Any ideas? BTW: trying to make the X100P channel 1 does not help either. ztcfg shows the channels, but asterisk will fail to load. Cheers, Willy - Original Message Follows - > Did you compile the zap and lipri and installed ? > > /HHA > > > app_dial.c:533 dial_exec: Unable to create channel of > > type 'ZAP' > > > > > ___ > 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 Newbie: TDM Development Kit
Hi!
I am a complete newbie at *asterisk.
I am somwhat stumped. I have *asterisk successfully running
on a FC-1 distro.
modprobe indicates that wcfxo and wcfxs are successfully
installed.
To get things started, I setup a simple SIP with two clients
(softphones) on my home LAN.
This works OK. I used the setup example (slightly modified)
from the Jon Todd article.
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html?page=1
and I found instructions on the X-lite phones. It appears
to work.
Then, I recieved the Digium development kit X100P and
TDM400P.
I added a line to my extensions as follows:
exten => _9.,1,Dial(ZAP/1/${EXTEN})
Figuring that anything which starts with a 9 should dial
outside the box.
Asterix console reports the following error:
app_dial.c:533 dial_exec: Unable to create channel of type
'ZAP'
Now, I probably need to do something to the zapata.conf
file, but I have no idea what.
Of course, the kit also has a TDM400P card. I have a phone
plugged into it, but (needless to say) there is no dial tone
or anything.
Help Please ...
willy
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