RE: [Asterisk-Users] Cisco Gateways
Hi, could you paste in some config examples and also share what you mean with 'little bugs'? THX, David Luyens -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Jones Sent: Tuesday, September 16, 2003 9:54 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Cisco Gateways Same here... Works great once you get the little bugs worked out. Brian. - Original Message - From: Michiel Betel [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 16, 2003 10:09 AM Subject: RE: [Asterisk-Users] Cisco Gateways I'm using cico's with SIP... And it works great :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Edward Gomez Sent: dinsdag 16 september 2003 15:52 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco Gateways Hi all, Just wondering if * can work with Cisco Gateways such as Cisco 2600/3600 routers or a VG200? -- Edward J. Gomez Director of Network Services ProxyMed, Inc 2555 Davie Road, Suite 110 Fort Lauderdale, Florida 33317 (954) 473-1001 x315 (954) 473-1656 FAX http://www.proxymed.com/ Confidential, unpublished property of ProxyMed, Inc. (c) copyright as of the date of this email. ProxyMed, Inc. CONFIDENTIALITY NOTICE: This e-mail message, including any attachments and files transmitted with it, are confidential and are intended solely for the use of the individual or entity to whom they are addressed. It may contain information that is privileged, confidential and exempt from disclosure under applicable laws. Moreover, this communication may contain the original sender's personal views and opinions, which do not necessarily reflect those of ProxyMed, Inc. . If the reader of this message is not the intended recipient, or the employee or agent responsible for delivering the message to the intended recipient, or if you have received this communication in error, please notify us immediately by return e-mail and delete the original message and any copies of it from your system. If you are not the intended recipient, be advised that you have received this e-mail in error, and that any unauthorized review, use, disclosure, distribution, forwarding, printing, or copying of this e-mail is strictly prohibited without our prior, written permission. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream Source?
have you more info on this free phone offer? please send it to me off the lest? senad -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Koehler Sent: 15 September 2003 23:08 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Grandstream Source? You get a Budgetone for free at Nikotel if you charge your account there with 100 bucks. The nikotel service works with *, even behind nat Tom (UnitedLayer) wrote: Anyone have a good source for BT-101 phones? I had a lead on some, but they've not materialized. I'm also interested in the ATA-286 (HandyTone) units as well. This is for my personal Asterisk/INOC-DBA setup, that has yet to materialize heh. --- Tom Sparks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MusicOnHold (MOH) silent on BudgeTone-100 only.
-= On Wed, 17 Sep 2003 11:29:07, Shaun Ewing [EMAIL PROTECTED] said: MoH works fine with my (local) Grandstream phones. It's just the direct-dialed music-only extension that does not. [...] Try something like exten = 6000,1,Answer exten = 6000,2,MusicOnHold Yes! You got it. That fixes it entirely. I've found that MoH won't be played unless the extension is answered first. -Shaun Thanks! This is certainly something to keep in mind and in the archives. The ATAs handle it differently, but it makes sense to Answer first. One less mystery. :) -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ISDN BRI active adapters with NT mode - any alternatives ?
On Tue, 2003-09-16 at 22:22, Klaus-Peter Junghanns wrote: Am Die, 2003-09-16 um 18.05 schrieb Louis-David Mitterrand: Now I have no idea if * supports plugging ISDN phones in the Diva. AFAIK it's not supported by chan_capi, but that may change. Yes, that may change. I will check with Eicon headquarters what the NT mode support in the BRI cards is about. Please keep us posted : I'm about to buy a Diva Server 4BRI and connecting ISDN phones (actually DECT base stations) is a critical capability; my project is a no go without it. And if somebody out there feels like sponsoring an Eicon BRI, so i can add support for it to chan_capi i wouldnt mind taking it ;-) (a place in the capi hall of fame will be yours) ;-) I had a hard time convincing management to fund the purchase of just one card for an experiment with Asterisk, so I guess I'm not going to be the one to sponsor one. But the development server is going to get a public IP address and an excellent symmetrical DSL connectivity, so if you want to play with that card I will gladly open an account with root privileges for you on that machine and do the local testing that you need. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm - IAX client for Win32
At 19:55 16-9-2003 -0500, you wrote: iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome. Well, this looks like a big improvement, but I cant seem to find the option to register at the asterisk server. Is it impossible, or am I missing it ? Would be a hefty requirement for real use, I think... Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help jeremy
* compiled from cvs, i am trying callip
phones in callmanager 3.2
10.17.0.2 is my callmanager
i noticed from network dumps that instead of
sending rtp to the ip phone,* sends it to 10.17.0.2!
thereby causing no audio from* to ip phone.
audio from ip phone to* is ok.
only callmanager calls fail. netmeeting works
ok...
here is the debug, thanks for any info
~kelvin
H323 debug enabled --
Executing Dial("SIP/kelvin-a8bc", "H323/[EMAIL PROTECTED]") in new
stack-- Making call to [EMAIL PROTECTED].
== New H.323 Connection created.
-- root is calling host [EMAIL PROTECTED]
-- Call token is
ip$localhost/18913 -- Call
reference is 18913 -- Called [EMAIL PROTECTED] us:
0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsReceiver us: 0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsTransmitter us: 0.0.0.0:18004them:
0.0.0.0:0info: 0.0.0.0:18004
=*= In CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsReceiver us: 0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsTransmitter us: 0.0.0.0:18004them:
0.0.0.0:0info: 0.0.0.0:18004
=*= In CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsReceiver us: 0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsTransmitter us: 0.0.0.0:18004them:
0.0.0.0:0info: 0.0.0.0:18004
=*= In CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsReceiver us: 0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsTransmitter us: 0.0.0.0:18004them:
0.0.0.0:0info: 0.0.0.0:18004
=*= In CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsReceiver us: 0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsTransmitter us: 0.0.0.0:18004them:
0.0.0.0:0info: 0.0.0.0:18004
=*= In CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsReceiver us: 0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsTransmitter us: 0.0.0.0:18004them:
0.0.0.0:0info: 0.0.0.0:18004
=*= In CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsReceiver us: 0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsTransmitter us: 0.0.0.0:18004them:
0.0.0.0:0info: 0.0.0.0:18004
=*= In CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsReceiver us: 0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsTransmitter us: 0.0.0.0:18004them:
0.0.0.0:0info: 0.0.0.0:18004
=*= In CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsReceiver us: 0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsTransmitter us: 0.0.0.0:18004them:
0.0.0.0:0info: 0.0.0.0:18004
=*= In CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsReceiver us: 0.0.0.0:18004them: 0.0.0.0:0info:
0.0.0.0:18004 =*= In
CreateRealTimeLogicalChannel for call
18913
-- externalIpAddress:
10.17.0.100
-- externalPort:
18004
-- SessionID:
1
-- Direction: IsTransmitter us: 0.0.0.0:18004them:
0.0.0.0:0info: 0.0.0.0:18004
=*= In
Re: [Asterisk-Users] Grandstream Source?
Senad Jordanovic wrote: have you more info on this free phone offer? please send it to me off the lest? Just as a totally wild guess, and call me crazy and amazingly intelligent for thinking of it, but how about looking at www.nikotel.com? I remain astonished by how many people need constant spoon feeding... -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call center design question
On Wed, 2003-09-17 at 03:50, Jean-Denis Girard wrote: Rich Adamson a crit : Would like to deploy * in a small help desk environment (five to ten people) using call queues and some sort of CTI interface to pop Remedy screen data in front of the help desk person receiving the call. Data to be popped would be based on CallerID. Anyone doing something similar? Anyone interfacing to an external Remedy system? Any reference sites that I could read/learn more of the requirements and/or 10,000 foot implementation? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I deployed a small call center using Gnophone as the screen data, together with dial + URL. Basically when the operator answers someone from the queue, an URL is pushed and displayed in Gnophone; this is quite simple as it is only web technology. The limitation is that no data is displayed until the called is transfered. Hello, I would like to create this kind of call center. Can you provide me more information about that ? Thx, Areski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help jeremy
chan_h323 doesn't currently inter operate with Call Manager, because I haven't been able to dedicate enough time to make native bridging work. Hell, maybe chan_skinny is the best way to interface CCM to Asterisk. Only if I had a non-production CCM to play with and more time. Jeremy Kelvin Chua wrote: * compiled from cvs, i am trying call ip phones in callmanager 3.2 10.17.0.2 is my callmanager i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2! thereby causing no audio from * to ip phone. audio from ip phone to * is ok. only callmanager calls fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin H323 debug enabled -- Executing Dial(SIP/kelvin-a8bc, H323/[EMAIL PROTECTED] mailto:H323/[EMAIL PROTECTED]) in new stack -- Making call to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]. == New H.323 Connection created. -- root is calling host [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -- Call token is ip$localhost/18913 -- Call reference is 18913 -- Called [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1
Re: [Asterisk-Users] help jeremy
FYI, people have reported that asterisk-oh323 works fine with CCM (haven't tested that myself). Michael. Kelvin Chua wrote: * compiled from cvs, i am trying call ip phones in callmanager 3.2 10.17.0.2 is my callmanager i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2! thereby causing no audio from * to ip phone. audio from ip phone to * is ok. only callmanager calls fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
- Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Caller ID Problems (WipeOut .) 2. Re: IAX, IAX2 and authenticatyion (Dan) 3. RE: 7206 as SIP-PSTN Gateway? (Abdul Hakeem) 4. Re: IAX, IAX2 and authenticatyion (Brancaleoni Matteo) 5. Re: Dect Phone (Tjardick van der Kraan) 6. Monitoring an active channel (Timothy Soos) 7. Re: asterisk and defunct perl procs (Rich Adamson) 8. Re: Caller ID Problems (Rich Adamson) 9. UK Suppliers (Angel Gabriel) 10. RE: UK Suppliers (Lee Redmayne) 11. How to test * ? (Angel Gabriel) 12. Re: IAX, IAX2 and authenticatyion ([EMAIL PROTECTED]) 13. Re: UK Suppliers (YO Internet Information) 14. Re: asterisk and defunct perl procs (Angel Gabriel) 15. Re: asterisk and defunct perl procs (Rich Adamson) 16. Re: Asterisk using a h323 gateway (Michael Manousos) --__--__-- Message: 1 From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Sat, 13 Sep 2003 06:41:43 + Subject: Re: [Asterisk-Users] Caller ID Problems Reply-To: [EMAIL PROTECTED] There are two things I can think of.. 1. You are not paying for CallerID support from your telco on that line.. Its is not always a standard feature.. 2. The CallerID that your telco provides is not compatible with the digium card and Asterisk.. Dear Asterisk User, I am trying to use a Digium FXO Card to get the callerid but fail. Asterisk version: Asterisk CVS-09/03/03-11:15:03 In my zapata.conf usecallerid=yes hidecallerid=no callwaitingcallerid=yes rxgain=3.0 txgain=3.0 ;callprogress=yes When I use my mobile (my mobile will show callerid) dial a call to the system Zap/1-1 channel. Then I use show channel zap/1-1 The callerid field show Caller ID: (N/A) Please help ... Anywhere I can check and anywhere I done wrong? Thanks, Randal -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze --__--__-- Message: 2 From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion Date: Sat, 13 Sep 2003 09:49:13 +0300 Organization: Personal Use Reply-To: [EMAIL PROTECTED] Hi Martin, - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, September 12, 2003 11:11 PM Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion IAX2 uses 4569 UDP port. How this port can be changed? There is no iax2.conf file... Dan --__--__-- Message: 3 From: Abdul Hakeem [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 7206 as SIP-PSTN Gateway? Date: Sat, 13 Sep 2003 08:21:40 +0100 Reply-To: [EMAIL PROTECTED] Hi, You need the PA-VFC-2TE1+ cards. It supports 60 calls for codecs such as G723 and 120 calls for G729a and b(with the addition of a PA-MCX card). Cheers, Abdul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Kane Sent: 12 September 2003 18:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7206 as SIP-PSTN Gateway? Also, don't limit yourself to Cisco. There are many vendors out there that make SIP trunking gateways... - Original Message - From: David C. Troy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 12, 2003 1:24 PM Subject: [Asterisk-Users] 7206 as SIP-PSTN Gateway? All, I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway. Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know which cards, if any, exist for a 7206VXR to act in a similar capacity, either as a T1/PRI, DS3, or POTS FXO/FXS? What other Cisco routers can act as SIP gateways today? Thanks, Dave = David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales ToadNet - Want to go fast?410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
Re: [Asterisk-Users] help jeremy
only for versions 0.5.1 version above this causes segmentation fault. i use 0.5.1 and it's ok, but it won't run in * cvs - Original Message - From: Michael Manousos [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 17, 2003 6:11 PM Subject: Re: [Asterisk-Users] help jeremy FYI, people have reported that asterisk-oh323 works fine with CCM (haven't tested that myself). Michael. Kelvin Chua wrote: * compiled from cvs, i am trying call ip phones in callmanager 3.2 10.17.0.2 is my callmanager i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2! thereby causing no audio from * to ip phone. audio from ip phone to * is ok. only callmanager calls fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help jeremy
ok thanks jerjer i'll take a peek at the chan_skinny ~kelvin - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 17, 2003 5:52 PM Subject: Re: [Asterisk-Users] help jeremy chan_h323 doesn't currently inter operate with Call Manager, because I haven't been able to dedicate enough time to make native bridging work. Hell, maybe chan_skinny is the best way to interface CCM to Asterisk. Only if I had a non-production CCM to play with and more time. Jeremy Kelvin Chua wrote: * compiled from cvs, i am trying call ip phones in callmanager 3.2 10.17.0.2 is my callmanager i noticed from network dumps that instead of sending rtp to the ip phone, * sends it to 10.17.0.2! thereby causing no audio from * to ip phone. audio from ip phone to * is ok. only callmanager calls fail. netmeeting works ok... here is the debug, thanks for any info ~kelvin H323 debug enabled -- Executing Dial(SIP/kelvin-a8bc, H323/[EMAIL PROTECTED] mailto:H323/[EMAIL PROTECTED]) in new stack -- Making call to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]. == New H.323 Connection created. -- root is calling host [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] -- Call token is ip$localhost/18913 -- Call reference is 18913 -- Called [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsReceiver us: 0.0.0.0:18004 them: 0.0.0.0:0 info: 0.0.0.0:18004 =*= In CreateRealTimeLogicalChannel for call 18913 -- externalIpAddress: 10.17.0.100 -- externalPort: 18004 -- SessionID: 1 -- Direction: IsTransmitter us: 0.0.0.0:18004 them: 0.0.0.0:0
Re: [Asterisk-Users] help jeremy
Kelvin Chua wrote: only for versions 0.5.1 version above this causes segmentation fault. More details on this? (backtrace, logs, configuration,...) i use 0.5.1 and it's ok, but it won't run in * cvs Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk using a h323 gateway
- Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. Re: Caller ID Problems (WipeOut .) 2. Re: IAX, IAX2 and authenticatyion (Dan) 3. RE: 7206 as SIP-PSTN Gateway? (Abdul Hakeem) 4. Re: IAX, IAX2 and authenticatyion (Brancaleoni Matteo) 5. Re: Dect Phone (Tjardick van der Kraan) 6. Monitoring an active channel (Timothy Soos) 7. Re: asterisk and defunct perl procs (Rich Adamson) 8. Re: Caller ID Problems (Rich Adamson) 9. UK Suppliers (Angel Gabriel) 10. RE: UK Suppliers (Lee Redmayne) 11. How to test * ? (Angel Gabriel) 12. Re: IAX, IAX2 and authenticatyion ([EMAIL PROTECTED]) 13. Re: UK Suppliers (YO Internet Information) 14. Re: asterisk and defunct perl procs (Angel Gabriel) 15. Re: asterisk and defunct perl procs (Rich Adamson) 16. Re: Asterisk using a h323 gateway (Michael Manousos) --__--__-- Message: 1 From: WipeOut . [EMAIL PROTECTED] To: [EMAIL PROTECTED] Date: Sat, 13 Sep 2003 06:41:43 + Subject: Re: [Asterisk-Users] Caller ID Problems Reply-To: [EMAIL PROTECTED] There are two things I can think of.. 1. You are not paying for CallerID support from your telco on that line.. Its is not always a standard feature.. 2. The CallerID that your telco provides is not compatible with the digium card and Asterisk.. Dear Asterisk User, I am trying to use a Digium FXO Card to get the callerid but fail. Asterisk version: Asterisk CVS-09/03/03-11:15:03 In my zapata.conf usecallerid=yes hidecallerid=no callwaitingcallerid=yes rxgain=3.0 txgain=3.0 ;callprogress=yes When I use my mobile (my mobile will show callerid) dial a call to the system Zap/1-1 channel. Then I use show channel zap/1-1 The callerid field show Caller ID: (N/A) Please help ... Anywhere I can check and anywhere I done wrong? Thanks, Randal -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze --__--__-- Message: 2 From: Dan [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion Date: Sat, 13 Sep 2003 09:49:13 +0300 Organization: Personal Use Reply-To: [EMAIL PROTECTED] Hi Martin, - Original Message - From: Martin Pycko [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Friday, September 12, 2003 11:11 PM Subject: Re: [Asterisk-Users] IAX, IAX2 and authenticatyion IAX2 uses 4569 UDP port. How this port can be changed? There is no iax2.conf file... Dan --__--__-- Message: 3 From: Abdul Hakeem [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] 7206 as SIP-PSTN Gateway? Date: Sat, 13 Sep 2003 08:21:40 +0100 Reply-To: [EMAIL PROTECTED] Hi, You need the PA-VFC-2TE1+ cards. It supports 60 calls for codecs such as G723 and 120 calls for G729a and b(with the addition of a PA-MCX card). Cheers, Abdul -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Kane Sent: 12 September 2003 18:30 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 7206 as SIP-PSTN Gateway? Also, don't limit yourself to Cisco. There are many vendors out there that make SIP trunking gateways... - Original Message - From: David C. Troy [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, September 12, 2003 1:24 PM Subject: [Asterisk-Users] 7206 as SIP-PSTN Gateway? All, I know you can use, say, a 2620 w/2 port FXO card as a SIP gateway. Clearly you can use the 5300, 5800, and MGX8850 too. Does anyone know which cards, if any, exist for a 7206VXR to act in a similar capacity, either as a T1/PRI, DS3, or POTS FXO/FXS? What other Cisco routers can act as SIP gateways today? Thanks, Dave = David C. Troy [EMAIL PROTECTED] 410-384-2500 Sales ToadNet - Want to go fast?410-544-1329 FAX 570 Ritchie Highway, Severna Park, MD 21146-2925 www.toad.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED]
RE: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs
This may just be me, but When replying to a message from a digest, it would be proper to remove all the context except that to which you are replying so as not to have to scroll an entire mile to see your reply. I know if I was the person you were replying to, I probably wouldn't scroll all the way through the other 15 messages just to see a reply. Just my .02, Sorry if I seem a bit irrational, just irritated. -Josh -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shimul Kanti Barua Sent: Wednesday, September 17, 2003 4:21 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: Asterisk-Users digest, Vol 1 #1279 - 16 msgs - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs BIG OLE SNIP Message: 16 Date: Sat, 13 Sep 2003 16:32:32 +0300 From: Michael Manousos [EMAIL PROTECTED] Organization: inAccess Networks To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Asterisk using a h323 gateway Reply-To: [EMAIL PROTECTED] Cerrajetto wrote: Hello: I am testing Asterisk with oh323. My question is: can Asterisk route some calls thru a second h323 gateway (a h323 - PSTN gw)? - Asterisk ip: 192.168.1.10 - h323-PSTN gw: 192.168.1.20 I've tried: exten = _9,1,Dial(OH323/192.1.1.20) or exten = _9,1,Dial(OH323/[EMAIL PROTECTED]) I guess that 192.1.1.20 is a typo, right? You will have to give more info in order to be able to find the problem. Try to set these params in oh323.conf file: wrapLibTraceLevel=3 libTraceLevel=3 libTraceFile=/tmp/trace.txt Rerun and send me the /tmp/trace.txt file, oh323.conf and the screen log (off-list). but it does not work at all. If my h323 client directly uses 192.168.1.20 as h323 gateway, the calls are routed to the PSTN perfectly. What is the correct way to route some calls from Asterisk to another h323 gateway? Thank you, Mark Michael. Hi Mark, Yes, it is possible. I have test it with Asterisk and oh323. We have routed some calls thru a second h323 gateway (like Vegastream and Cirilium). Following is the configuration: ; Vegastream exten = _01XX,1,Dial(OH323/[EMAIL PROTECTED]) ; Crilium - exten = _9XX,1,Dial(OH323/[EMAIL PROTECTED]) Shimul --__--__-- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk using a h323 gateway
Why do I need to read all the other stuff just to get to a 3 liner ? On Wednesday 17 September 2003 12:58 pm, Shimul Kanti Barua wrote: - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs ... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] using pci modem cards as fxs/fxo ports in *
Hi all, forgive the question but is it possible to use PCI modem cards (aka winmodem's) as FXO/FXS ports in * ? what about external modems like the USR Sportsters? Thanks in advance, Bryan. Bryan Nolen Lead Developer http://Arc.Net.AU http://cdonline.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Aleatori PSTN number with SIP.
Hi everybody, Now I'm using SJphone on a win2k client an* as proxy SIP andGW to PSTN. I have doing some test, but I have the following question. It's possibles to make calls to external PSTN numbers without define an extension to make the call I will try to explain-me better. I have done some calls like sip:[EMAIL PROTECTED], where in extensions.conf there are an extension like this: exten=xisco,1,Answer exten=xisco,2,Dial(Zap/g1/definened number for me) I want to make a call like sip:my phonenumber or other number@213.229.160.218 without define the phone number anywhere. It is possible, how can I do it Thks a lot.
Re: [Asterisk-Users] LineJack + Asterisk HELP!
On Tue, 16 Sep 2003 12:53:18 -0300, Bartosz Jozwiak wrote Hello, Thanks very much for help. To install driver for LineJack I need kernel source. I have debian, and I installed from apt-get install kernel- source.2.4.20 but while it make ./configure it still asks me for the kernel source. What can be wrong ? -- Bart hi why not download plain kernel source? anyway, debian kernel-source packages contain only the kernel in .tar.bz2 format (or was it .tar.gz?), named /usr/src/kernel-source-*. you need to unpack it manually and make a symlink to /usr/src/linux probably. btw, make sure you're running the kernel you're compiling the driver for. hth, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LineJack + Asterisk HELP!
Hello, I have kernel-source-2.4.20.tar.gz and I untar this on. Should I try it once again with tar.bz2 ? I am ranning the same kernel for sure. - Original Message - From: Grzegorz Nosek [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 17, 2003 8:55 AM Subject: Re: [Asterisk-Users] LineJack + Asterisk HELP! On Tue, 16 Sep 2003 12:53:18 -0300, Bartosz Jozwiak wrote Hello, Thanks very much for help. To install driver for LineJack I need kernel source. I have debian, and I installed from apt-get install kernel- source.2.4.20 but while it make ./configure it still asks me for the kernel source. What can be wrong ? -- Bart hi why not download plain kernel source? anyway, debian kernel-source packages contain only the kernel in .tar.bz2 format (or was it .tar.gz?), named /usr/src/kernel-source-*. you need to unpack it manually and make a symlink to /usr/src/linux probably. btw, make sure you're running the kernel you're compiling the driver for. hth, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] using pci modem cards as fxs/fxo ports in *
Bryan Nolen wrote: forgive the question but is it possible to use PCI modem cards (aka winmodem's) as FXO/FXS ports in * ? what about external modems like the USR Sportsters? Methinks this needs to go in an FAQ, and the FAQ needs to be linked to from the mailing list signup page/confirmation e-mails. Long answer: Search the list archives/Google. Short answer: Nope. Lack of voice duplexing tends to scupper you. -- Alastair Maw [EMAIL PROTECTED] MX Telecom - Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LineJack + Asterisk HELP!
Steps what I did: I have install the kernel source the some one ofcourse of my system kernel. I have untar in to /usr/src/linux/ I made make menuconfig I apply linux telephony I made make dep Everything went ok. Then I made in ixj ./configure was ok then i mad make It gave me some error that files did not exist. So I change in /usr/src/linux/include/linux/modversions.h for correct folder. then make went ok Then make install and i got this: depmod: *** Unresolved symbols in /lib/modules/2.4.20/kernel/drivers/telephony/ixj.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/tor2.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/wcfxo.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/wcfxs.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/wct1xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/wct4xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/wcusb.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/ztd-eth.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/ztdynamic.o the rest were ok only these were looking strange to me. then I made modprobe ixj and got this: /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol try_inc_mod_count_Rsmp_e6105b23 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol register_chrdev_Rsmp_63ef0035 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol request_module_Rsmp_27e4dc04 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol unregister_chrdev_Rsmp_c192d491 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol sprintf_Rsmp_1d26aa98 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol printk_Rsmp_1b7d4074 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol panic_Rsmp_01075bf0 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: Hint: You are trying to load a module without a GPL compatible license and it has unresolved symbols. Contact the module supplier for assistance, only they can help you. /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o failed /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod ixj failed Strange. - Original Message - From: [EMAIL PROTECTED] To: Bartosz Jozwiak [EMAIL PROTECTED] Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Wednesday, September 17, 2003 1:06 AM Subject: Re: your mail Please try to tell me exactly what steps you did, and I will try to help you. It seems to be a non-asterisk issue so you can just email me directly. Please use a subject line or the spambouncer may not like it. Regards F On Tue, 16 Sep 2003, Bartosz Jozwiak wrote: Hello, I made install. Why I am getting this. My linux is Debian. -- Hi Looks like you did not do a make install after compiling the drivers, and it is still loading the stock kernel ixj. Please try doing a make install in the ixj-x.x.x source directory. Hope that helps On Tue, 16 Sep 2003, Bartosz Jozwiak wrote: Yes I fixed it thanks. But I have another problem. I am not so good with linux... so sorry If I am irritating... this is what i got: bmtst:/usr/src/ixj-1.2.1# modprobe ixj /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol try_inc_mod_count_Rsmp_e6105b23 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol register_chrdev_Rsmp_63ef0035 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol request_module_Rsmp_27e4dc04 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol unregister_chrdev_Rsmp_c192d491 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol sprintf_Rsmp_1d26aa98 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol printk_Rsmp_1b7d4074 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol panic_Rsmp_01075bf0 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: Hint: You are trying to load a module without a GPL compatible license and it has unresolved symbols. Contact the module supplier for assistance, only they can help you. /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o failed /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod ixj failed What can I do about it ? - Original Message - From: Daryl G. Jurbala [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, September 16,
Re: [Asterisk-Users] LineJack + Asterisk HELP!
I got the same. When I make modprobe phonedev i got the same thing: /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol try_inc_mod_count_Rsmp_e6105b23 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol register_chrdev_Rsmp_63ef0035 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol request_module_Rsmp_27e4dc04 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol unregister_chrdev_Rsmp_c192d491 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol sprintf_Rsmp_1d26aa98 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol printk_Rsmp_1b7d4074 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol panic_Rsmp_01075bf0 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: Hint: You are trying to load a module without a GPL compatible license and it has unresolved symbols. Contact the module supplier for assistance, only they can help you. /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o failed /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod phonedev failed - Original Message - From: [EMAIL PROTECTED] To: Bartosz Jozwiak [EMAIL PROTECTED] Sent: Wednesday, September 17, 2003 5:33 PM Subject: Re: [Asterisk-Users] LineJack + Asterisk HELP! make: modprobe phonedev modprobe ixj On Wed, 17 Sep 2003, Bartosz Jozwiak wrote: Steps what I did: I have install the kernel source the some one ofcourse of my system kernel. I have untar in to /usr/src/linux/ I made make menuconfig I apply linux telephony I made make dep Everything went ok. Then I made in ixj ./configure was ok then i mad make It gave me some error that files did not exist. So I change in /usr/src/linux/include/linux/modversions.h for correct folder. then make went ok Then make install and i got this: depmod: *** Unresolved symbols in /lib/modules/2.4.20/kernel/drivers/telephony/ixj.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/tor2.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/torisa.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/wcfxo.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/wcfxs.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/wct1xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/wct4xxp.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/wcusb.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/ztd-eth.o depmod: *** Unresolved symbols in /lib/modules/2.4.20/misc/ztdynamic.o the rest were ok only these were looking strange to me. then I made modprobe ixj and got this: /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol try_inc_mod_count_Rsmp_e6105b23 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol register_chrdev_Rsmp_63ef0035 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol request_module_Rsmp_27e4dc04 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol unregister_chrdev_Rsmp_c192d491 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol sprintf_Rsmp_1d26aa98 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol printk_Rsmp_1b7d4074 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: unresolved symbol panic_Rsmp_01075bf0 /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: Hint: You are trying to load a module without a GPL compatible license and it has unresolved symbols. Contact the module supplier for assistance, only they can help you. /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o failed /lib/modules/2.4.20/kernel/drivers/telephony/phonedev.o: insmod ixj failed Strange. - Original Message - From: [EMAIL PROTECTED] To: Bartosz Jozwiak [EMAIL PROTECTED] Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED] Sent: Wednesday, September 17, 2003 1:06 AM Subject: Re: your mail Please try to tell me exactly what steps you did, and I will try to help you. It seems to be a non-asterisk issue so you can just email me directly. Please use a subject line or the spambouncer may not like it. Regards F On Tue, 16 Sep 2003, Bartosz Jozwiak wrote: Hello, I made install. Why I am getting this. My linux is Debian. -- Hi Looks like you did not do a make install after compiling the drivers, and it is still loading the stock kernel ixj. Please try doing a make install
Re: [Asterisk-Users] NEW Asterisk Security vulnerability report ...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Lubomir Christov wrote: | Hello, | | There is a new asterisk vulnerability report at this address: | | http://www.securiteam.com/unixfocus/5HP0H1PB5S.html | | This is the second security report regarding asterisk for 8 days | (http://www.securiteam.com/securitynews/5LP0720B5G.html) | | Both fixes was reported and fixed silently. | | My question is: Is it possible in the future such a security problems to | be reported in this mailing list or some other security related list? | I would really like to see Asterisk security fixes posted to BugTraq, as that is where I monitor for vulnerabilities in my boxes. - -- Leif Madsen. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/aGm16gq3eQ0gpNURAohaAKCg9RL93co6fAfoxJA0fgrSsor0hgCdE1y1 C5sAMippFb6fK7q0xiik6O4= =eL29 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A WORKING EXAMPLE
Hello! I've looked at the reference examples they are all for SIP. I have two X100p and a TDM400P. Can someone send me a working example so I can receive calls and make them. I'm stuck at first base. [I'm using standard phones - not SIP] Help please! Thanks, Bill Flood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NEW Asterisk Security vulnerability report ...
There is a new asterisk vulnerability report at this address: http://www.securiteam.com/unixfocus/5HP0H1PB5S.html This is the second security report regarding asterisk for 8 days (http://www.securiteam.com/securitynews/5LP0720B5G.html) Both fixes was reported and fixed silently. My question is: Is it possible in the future such a security problems to be reported in this mailing list or some other security related list? Of course, this particular bug is likely only going to affect a small subset of people for the following reasons: a) Don't accept VoIP from untrusted sources b) Their telco doesn't permit untrusted source to spoof callerid c) They don't use the SQL CDR recording d) Without actually looking into it, what is the maxlength of callerid anyway? I'm also wondering why it took so long for this bug to be fixed? Also, the list should be notified once the fix is in CVS (which should be when bugtraq etc is notified) Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WebVoiceMail forward message error
Hi all: I have a problem in VoiceMail application when I forward a message to another extension. The error is: Software error: Invalid old MessageBR For help, please send mail to the webmaster ([EMAIL PROTECTED]), giving this error message and the time and date of the error. What can I do? Regards, Gus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan_h323/g729 - X100P connecting to non-Digium Partner
Since then I couldn't test it, but now I installed EtheReal last version with h323 support. Did some calls and perceived that the call is being cut after the Master/Slave negotiation. Asterisk is sending an EndSession as you can see in the file attached. If the list doesn't allow attachments, the same file can be found at http://isamarmaia.org/packets.pak BTW, I didn't find the patch you mentioned. Could you gimme its URL? Thanks a lot, Isamar Maia On Wed, 27 Aug 2003 [EMAIL PROTECTED] wrote: Hi The endpoint seems to be running Radvision h323 stack, and I know chan_h323 works with Radvision, there could be a couple of reasons!! 1) You dont have G729A in the capabilities of remote endpoint 2) The packetization interval is way off The best way would be to run ethereal or dump323 and see what is being negotiated. Also try to use fastConnect on both sides and force same packetization, (you can use my patch posted a couple of days ago to force packetization interval in G729 in chan_h323) Isamar Said I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2 HSMU RAD: cmHookSend(masterSlaveDeterminationAck) 11- HSMU 0 Remote capabilities list: 0- HSMU 0 [1] g729AnnexA: Audio Receive 0- HSMU 0 Try matching local element: 0- HSMU 0 [1] g7231: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [2] g729: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [3] g711Ulaw64k: Audio Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [4] t38fax: Data Receive and Transmit 0- HSMU 0 Try matching local element: 0- HSMU 0 [5] g729: Audio Receive and Transmit 0- HSMU 3 HSMU 2: Capabilities: NO MATCH FOUND! 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE - error release 1- HSMU 2 HSMU 2: abort timer in state H245 WAIT COMPLETE 0- HSSM 2 HSMU 2: SM H245 WAIT COMPLETE == RELEASING 10- RADH 2 HSMU RAD: cmHookSend(endSessionCommand) 3- RAD 2 HSMU 2: cmEvCallControlStateChanged(cmControlStateTransportDisconnected, - ) Anybody has any idea? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users packets.pak Description: Binary data
[Asterisk-Users] Sip call waiting
Hi folks, As none of the SIP softphones that I tested can disable more than one incoming call, I decided to implement it by software ;-) I'm attaching a patch that does it. To make it work, modify your sip.conf file and include callwaiting=[0|1] at the general section, or for each peer that you wish to control. Please note that I haven't tested it too much, and my source tree is quite old, so I'm not sure if this patch will apply to the current CVS. Let me know if you find something wrong asap, as this goes into production tomorrow ! Best regards, PauloHM sipcallwaiting.diff Description: Binary data
Re: [Asterisk-Users] Hangups after voicemail
Do you have silence in the channel when the remote user hangs up or busy
tone ?
If you have silence you can use maxsilence=x_seconds in voicemail.conf
with
Voicemail2 application and that will make sure the calls are hanged up
after x_seconds of silence in the channel.
If you have busy tone then use the busydetect=yes in zapata.conf.
You can also limit the length of the voicemail message with
maxmessage=x_seconds in the voicemail.conf
regards
Martin
On Tue, 16 Sep 2003, Christian Hecimovic wrote:
Hi,
Try as I might, I can't get hangups detected on a Zap channel with loop start
lines. So, after someone leaves a voicemail and then hangs up, Asterisk
doesn't know it, exits VoicemailMain2, and loops back to the corporate
greeting, tying up the line even though the outside caller has hung up.
Therefore, I've added the following hideous hack - er, code - to voicemail2.c.
It starts right after the call to play_and_record() in leave_voicemail().
if (res != '#' chan != NULL !strncmp(chan-name, Zap, 3)) {
/* Hang up the Zap channel only */
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}
Obviously, it hangs up the channel after the voicemail has been recorded, if
the # key wasn't pressed, if the channel still exists, and if it's a Zap
channel. I couldn't see a way to do this with AGI.
Question: is this safe? I used a soft hangup because the channel is controlled
by another thread. I also modified channel.c so that ast_channel_free() sets
chan to NULL after it's freed, just in case. Is there anything else I should
be aware of? The code seems to work in my testing, resulting in a proper
hangup right after the voicemail has been recorded. I'm not up on my Asterisk
internals, so I'm not totally confident about this.
Thanks,
Chris
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Re: [Asterisk-Users] calls terminating abnormally
Can you send a pri debug span span_no trace ? Or do you have an analog T1/E1 ? regards Martin On Wed, 17 Sep 2003, denzel-infotechs wrote: hi! I've got a asterisk system running with around 50 per calls per minute. I've connected * to internal pabx and outside telecom using E1 (ISDN pris). Sometimes calls disconect abnormally. Is this something we have to live with or is it a bug in CVS code ? denzel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] A WORKING EXAMPLE
I've looked at the reference examples they are all for SIP. I have two
X100p and a TDM400P. Can someone send me a working example so I can
receive calls and make them. I'm stuck at first base. [I'm using standard
phones - not SIP] Help please!
I just implemented two X100p cards, but not the TDM card. Here's some steps
that I used in a lab environment, but keep in mind I'm also new to this so
I might have missed a couple of steps.
1. Use the two page document that came with the X100p cards and do everything
mentioned in that document.
2. Execute a /sbin/modprobe wcfxo as root (assuming you're RedHat)
3. Modify or create a /etc/zaptel.conf file and put only:
fxsks=1-2
loadzone=us
in it. The 1-2 indicates a range of x100p cards (eg, #1 and #2), and it is
configuring the cards as FXS (attaching to an incoming pots line).
4. Execute /usr/src/zaptel/ztcfg -vv as root.
Note: the above steps are installing the linux drivers, etc, getting ready
for asterisk to use them.
5. In the /etc/asterisk/extensions.conf file, put:
[from-sip]
ignorepat = 9
exten = _9X.,1,Dial,Zap/1/${EXTEN:1}
The above says... when an * extension dials 9, drop that digit and send
all remaining digits out the Zap/1 (first x100p) interface.
6. In the same /etc/asterisk/extensions.conf file, towards the top put:
[globals]
PHONE1=SIP/3000
PHONE2=SIP/3001
towards the bottom of the file, put:
[inbound-bus]
exten = s,1,Dial(${PHONE1}${PHONE2},15)
; exten = s,2,Wait,2
; exten = s,2,Voicemail,u3001
; exten = s,102,Voicemail,b3001
The above says... were defining two global variables (PHONE1 and PHONE2)
and setting their values to extension 3000 and 3001 (these are assuming
sip extensions, regardless of whether they are sip phones or ata186).
Then, when a call comes in to the [inbound-bus] context, it will ring
both extensions at the same time for up to 15 seconds. If you uncomment
the three lines shown, the call will roll over to Voicemail box 3001 if
the call is unanswered, or, to Voicemail box 3001 if it is busy.
7. In the /etc/asterisk/zapata.conf file, towards the bottom put:
context=inbound-bus; this is the context that appears in extensions.conf
switchtype=national
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
pickupgroup=1
immediate=no
callprogress=no
musiconhold=default
channel = 1; this is the x100p #1 or #2 card (#1 is specified here).
In effect, step 7 configures and receives the incoming pots calls (for one
line, duplicate it for the second x100p card), and hands the incoming call
to the inbound-bus context in extensions.conf, which then rings whatever
extension you've configured in step 6.
For everyone else reading this on the list that are more experienced then
I, feel free to add/change/delete steps as technically necessary. No
pride of authorship here. ;)
Rich
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[Asterisk-Users] chan_h323 as a gatekeeper?
hi IIRC, Jeremy once said that chan_h323 could be used as a gatekeeper but perhaps lacking a few features as compared to gnugk. Is this possible? I have some dlink DPH-100H phoes here for testing, but they require a gatekeeper, and if I can do it, I'd love to keep gnugk out of this. thanks roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phpconfig is out in CVS
I gives you a way to see extensions called out and sorted and a bit more friendly than vi. Its also a framework to get wizzards added to. like create VM box etc... Dave [EMAIL PROTECTED] 9/15/2003 4:45:58 PM I am sorry if I am missing something but!!, How is this any different than using a text editor. What does it give you that (for example) using vi on SSH/Telnet Java Applet that comes with WebMin doesnt give you. In some ways it is actually very limiting. Sorry, but had to ask On Sat, 13 Sep 2003, Darren Poulson wrote: Nice one! Took all of about 30 seconds to install, including downloading from the net. Just got the latest CVS and copied it into the web folder and opened up konqueror. Everything seems to be working fine. Off to do some testing of it now. Cheers, Darren. On Friday 12 Sep 2003 11:34 am, Peter Pauly wrote: On Thu, Sep 11, 2003 at 10:12:50PM -0600, Dave Packham wrote: nope when I click on something on the left I get a FQDN not just the pne you had Hmmm. Further info: it works with Microsoft Internet Explorer. It does not work with Mozilla 1.4 under Linux. It also does work with Mozilla Firebird under Windows. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phpconfig is out in CVS
The Linux problem is a Netscape problem. NS currently has a bug in the find code that prevents it from doing a find() in a textbox. NS knows about this and it will be fixed soon. Firebird works and some other non NS dependant browsers work too. if you can think of a way to do a find in a textfield without find on NS lemme know and Ill add that Dave [EMAIL PROTECTED] 9/15/2003 4:23:51 PM And also it require IE for search, which a linux admin will probably not gonna have (I dont use M$ products). BTW there are ways to do that on Mozilla too. On Thu, 11 Sep 2003, Peter Pauly wrote: On Thu, Sep 11, 2003 at 07:57:58PM -0600, Dave Packham wrote: I have put my phpconfig stuff out into the Digium CVS tree. Project name is phpconfig. see it at http://rads.netcom.utah.edu/phpconfig/phpconfig.php Looks cool, but the links don't work on the left. It wants http://phpconfig/phpconfig.php? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: call center design question
Rich Adamson a écrit : Would like to deploy * in a small help desk environment (five to ten people) using call queues and some sort of CTI interface to pop Remedy screen data in front of the help desk person receiving the call. Data to be popped would be based on CallerID. Anyone doing something similar? Anyone interfacing to an external Remedy system? Any reference sites that I could read/learn more of the requirements and/or 10,000 foot implementation? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users I deployed a small call center using Gnophone as the screen data, together with dial + URL. Basically when the operator answers someone from the queue, an URL is pushed and displayed in Gnophone; this is quite simple as it is only web technology. The limitation is that no data is displayed until the called is transfered. I would really like to have more info about this! Is it possible? BTW Gnophone uses IAX. Does anybody knows if there is a good IAX softphone for Windows? Ben ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangups after voicemail
Hi Wade,
Yes, my zapata.conf looks like this:
group = 1
context = incoming
signalling = fxs_ks
echocancel = yes
echocancelwhenbridged = yes
channel = 1-2
So they are configured as kewlstart.
Thanks,
Chris
On Tuesday 16 September 2003 16:53, Wade J. Weppler wrote:
Have you tried using kewlstart instead? Your loopstart lines might be
configured for kewlstart (forward disconnect supervision).
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 16, 2003 7:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Hangups after voicemail
Hi,
Try as I might, I can't get hangups detected on a Zap channel with
loop
start
lines. So, after someone leaves a voicemail and then hangs up,
Asterisk
doesn't know it, exits VoicemailMain2, and loops back to the corporate
greeting, tying up the line even though the outside caller has hung
up.
Therefore, I've added the following hideous hack - er, code - to
voicemail2.c.
It starts right after the call to play_and_record() in
leave_voicemail().
if (res != '#' chan != NULL !strncmp(chan-name, Zap, 3)) {
/* Hang up the Zap channel only */
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}
Obviously, it hangs up the channel after the voicemail has been
recorded,
if
the # key wasn't pressed, if the channel still exists, and if it's a
Zap
channel. I couldn't see a way to do this with AGI.
Question: is this safe? I used a soft hangup because the channel is
controlled
by another thread. I also modified channel.c so that
ast_channel_free()
sets
chan to NULL after it's freed, just in case. Is there anything else I
should
be aware of? The code seems to work in my testing, resulting in a
proper
hangup right after the voicemail has been recorded. I'm not up on my
Asterisk
internals, so I'm not totally confident about this.
Thanks,
Chris
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RE: [Asterisk-Users] Hangups after voicemail
Silencethreshold of 256 sounds a bit high...
You can also add a timeout extension to just hangup the line:
exten = t,1,Hangup
Without using Kewlstart, there isn't anyway for Asterisk to know that
the line has been disconnected, so you'll have to use the timeouts.
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 11:59 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hangups after voicemail
Hi Martin,
Yes, silence detection in voicemail is working. I am using Voicemail2
with
the
silencethreshold set to 256. However, the line doesn't hang up after
the
silence is detected; instead, Voicemail2 exits after recording the
voicemail
correctly, and Asterisk loops back into the main menu as if the # key
was
pressed because the channel is still alive. Then it times out after 15
seconds, as you can see below.
From extensions.conf:
[incoming]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,BackGround(corp_greeting)
include = locals
include = errors
The locals context consists of macros which look like this:
exten = s,1,Playback(transfer,skip)
exten = s,2,Dial(${ARG2},20)
exten = s,3,Voicemail2(u${ARG1})
exten = s,4,Goto(incoming,s,1)
exten = s,103,Voicemail2(b${ARG1})
exten = s,104,Goto(incoming,s,1)
So after a voicemail is left, there is a Goto back into the incoming
context.
It all works great, except for when the line gets tied up by the
DigitTimeout
and ResponseTimeout bits when hangups aren't detected.
I've tried using BUSYDETECT_MARTIN with busydetect=yes and it didn't
work.
The
channel stays up after the outside caller hangs up.
Since all of our inside phones are SIP lines, there is no problem
detecting
hangups when a voice conversation is taking place, since Asterisk
obviously
detects SIP hangups correctly whether it's SIP to SIP or SIP to
outside
line.
The problem is really only when outside callers leave voicemail.
Thanks,
Chris
On Wednesday 17 September 2003 08:09, Martin Pycko wrote:
Do you have silence in the channel when the remote user hangs up or
busy
tone ?
If you have silence you can use maxsilence=x_seconds in
voicemail.conf
with
Voicemail2 application and that will make sure the calls are hanged
up
after x_seconds of silence in the channel.
If you have busy tone then use the busydetect=yes in zapata.conf.
You can also limit the length of the voicemail message with
maxmessage=x_seconds in the voicemail.conf
regards
Martin
On Tue, 16 Sep 2003, Christian Hecimovic wrote:
Hi,
Try as I might, I can't get hangups detected on a Zap channel with
loop
start lines. So, after someone leaves a voicemail and then hangs
up,
Asterisk doesn't know it, exits VoicemailMain2, and loops back to
the
corporate greeting, tying up the line even though the outside
caller
has
hung up.
Therefore, I've added the following hideous hack - er, code - to
voicemail2.c. It starts right after the call to play_and_record()
in
leave_voicemail().
if (res != '#' chan != NULL !strncmp(chan-name, Zap, 3))
{
/* Hang up the Zap channel only */
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}
Obviously, it hangs up the channel after the voicemail has been
recorded,
if the # key wasn't pressed, if the channel still exists, and if
it's
a
Zap channel. I couldn't see a way to do this with AGI.
Question: is this safe? I used a soft hangup because the channel
is
controlled by another thread. I also modified channel.c so that
ast_channel_free() sets chan to NULL after it's freed, just in
case.
Is
there anything else I should be aware of? The code seems to work
in my
testing, resulting in a proper hangup right after the voicemail
has
been
recorded. I'm not up on my Asterisk internals, so I'm not totally
confident about this.
Thanks,
Chris
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[Asterisk-Users] documentation?
Been learning * now for a couple of weeks and have all basic features running including VM, MoH, FX lines, iaxtel, and FWD. However, I seem to be lacking documentation on a lot of technical things and am wondering if I overlooked something that is obvious to others. (I do have the Handbook, have been doing a fair amount of google searches, and read the README.* files.) Examples, Where should I have learned that *8# is the call pickup dialing sequence? Other then *8#, are there other preprogrammed sequences (I assume there is) in a stock * implementation? Where should I have learned about the t and r in: exten = s,1,Dial(SIP/3000,20,tr) and all the other possible options in various * config statements? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail2 and time stamps
Created new message, not a response with new Subject: line -- check Plain text mode set in Outlook -- check Good day all, I just recently upgraded to 0.5.0 and subsequent CVS releases. After the upgrade, I've noticed that voicemail2 has differing opinions on what time it is, assume this is because I haven't set the TZ per user in voicemail.conf. Anyone have an example as to how to configure per user? Cheers, --- Gavin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hangups after voicemail
Then it sounds like your Telco isn't giving you kewlstart signaling.
This is by far the most reliable method of telling asterisk that the
line has been disconnected. Trying asking your Telco if they can supply
you with Kewlstart or Forward Disconnect Supervision on your line.
Basically, all this does is momentarily reverse the polarity on the line
to indicate that the line has been disconnected. The Zaptel FXO devices
detect this condition to indicate to Asterisk that the line has been
disconnected.
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 12:01 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hangups after voicemail
Hi Wade,
Yes, my zapata.conf looks like this:
group = 1
context = incoming
signalling = fxs_ks
echocancel = yes
echocancelwhenbridged = yes
channel = 1-2
So they are configured as kewlstart.
Thanks,
Chris
On Tuesday 16 September 2003 16:53, Wade J. Weppler wrote:
Have you tried using kewlstart instead? Your loopstart lines might
be
configured for kewlstart (forward disconnect supervision).
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 16, 2003 7:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Hangups after voicemail
Hi,
Try as I might, I can't get hangups detected on a Zap channel with
loop
start
lines. So, after someone leaves a voicemail and then hangs up,
Asterisk
doesn't know it, exits VoicemailMain2, and loops back to the
corporate
greeting, tying up the line even though the outside caller has
hung
up.
Therefore, I've added the following hideous hack - er, code - to
voicemail2.c.
It starts right after the call to play_and_record() in
leave_voicemail().
if (res != '#' chan != NULL !strncmp(chan-name, Zap, 3))
{
/* Hang up the Zap channel only */
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}
Obviously, it hangs up the channel after the voicemail has been
recorded,
if
the # key wasn't pressed, if the channel still exists, and if it's
a
Zap
channel. I couldn't see a way to do this with AGI.
Question: is this safe? I used a soft hangup because the channel
is
controlled
by another thread. I also modified channel.c so that
ast_channel_free()
sets
chan to NULL after it's freed, just in case. Is there anything
else I
should
be aware of? The code seems to work in my testing, resulting in a
proper
hangup right after the voicemail has been recorded. I'm not up on
my
Asterisk
internals, so I'm not totally confident about this.
Thanks,
Chris
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[Asterisk-Users] Configuration for Asterisk with Cisco Router FXO
Since I see so many questions about this, and could not find a concise answer when I was looking for the same thing... Here is an example showing how to configure communications between Asterisk and a Cisco 2600 router with an FXO card in it. http://www.tape.net/~gerry/asterisk/cisco26x0.html Comments and suggestions are welcome. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Adpcm, 6KHz codec
Title: RE: [Asterisk-Users] Adpcm, 6KHz codec I am positive, 4 bits per sample, 6000 Hz. This is a default play/record setting for the older Dialogic R4 API and we need to play zillions (sic!) of files (messages) recorded this way. Conversion issues: - expensive - resampling quality - storage - application changes - etc. Would be real nice and useful to have this codec. Thank you. Alex Zarubin Webley Systems, Inc. -Original Message- From: Mark Spencer [mailto:[EMAIL PROTECTED]] Sent: Tuesday, September 16, 2003 11:23 PM To: '[EMAIL PROTECTED]' Subject: RE: [Asterisk-Users] Adpcm, 6KHz codec What I need is adpcm algorithm (4 bits per sample) with 6 KHz sampling rate (6000 samples per second). This is 24kbps. Are you sure you're not thinking of 3 bits per sample 8000 Hz ADPCM (also 2400kbps)? Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sample paging config
Hi, Can someone please post a sample config (oss.conf, extensions.conf, etc.) of what is necessary to use the soundcard in the Asterisk server to do overhead paging? Thank you. Travis Microserv  ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangups after voicemail
Yes, I had to set it high, otherwise it didn't work right...I'll fiddle with
it a bit.
Timeouts aren't really an option, because if the caller presses # after
leaving a voicemail then they should be popped back into the main menu. If I
could check DTMF signals from extensions.conf, then this would indeed work.
Basically, the logic after Voicemail2 exits should be
if (last key was not # and the channel is still alive)
hangup
else
goto the main menu
The only way I could do this was by modifying voicemail2.c. So, back to my
main question: are there any problems with this? I'm most concerned about
memory issues. Should I be freeing something first, making another cleanup
function call, etc.?
On Wednesday 17 September 2003 09:08, Wade J. Weppler wrote:
Silencethreshold of 256 sounds a bit high...
You can also add a timeout extension to just hangup the line:
exten = t,1,Hangup
Without using Kewlstart, there isn't anyway for Asterisk to know that
the line has been disconnected, so you'll have to use the timeouts.
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 11:59 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hangups after voicemail
Hi Martin,
Yes, silence detection in voicemail is working. I am using Voicemail2
with
the
silencethreshold set to 256. However, the line doesn't hang up after
the
silence is detected; instead, Voicemail2 exits after recording the
voicemail
correctly, and Asterisk loops back into the main menu as if the # key
was
pressed because the channel is still alive. Then it times out after 15
seconds, as you can see below.
From extensions.conf:
[incoming]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,BackGround(corp_greeting)
include = locals
include = errors
The locals context consists of macros which look like this:
exten = s,1,Playback(transfer,skip)
exten = s,2,Dial(${ARG2},20)
exten = s,3,Voicemail2(u${ARG1})
exten = s,4,Goto(incoming,s,1)
exten = s,103,Voicemail2(b${ARG1})
exten = s,104,Goto(incoming,s,1)
So after a voicemail is left, there is a Goto back into the incoming
context.
It all works great, except for when the line gets tied up by the
DigitTimeout
and ResponseTimeout bits when hangups aren't detected.
I've tried using BUSYDETECT_MARTIN with busydetect=yes and it didn't
work.
The
channel stays up after the outside caller hangs up.
Since all of our inside phones are SIP lines, there is no problem
detecting
hangups when a voice conversation is taking place, since Asterisk
obviously
detects SIP hangups correctly whether it's SIP to SIP or SIP to
outside
line.
The problem is really only when outside callers leave voicemail.
Thanks,
Chris
On Wednesday 17 September 2003 08:09, Martin Pycko wrote:
Do you have silence in the channel when the remote user hangs up or
busy
tone ?
If you have silence you can use maxsilence=x_seconds in
voicemail.conf
with
Voicemail2 application and that will make sure the calls are hanged
up
after x_seconds of silence in the channel.
If you have busy tone then use the busydetect=yes in zapata.conf.
You can also limit the length of the voicemail message with
maxmessage=x_seconds in the voicemail.conf
regards
Martin
On Tue, 16 Sep 2003, Christian Hecimovic wrote:
Hi,
Try as I might, I can't get hangups detected on a Zap channel with
loop
start lines. So, after someone leaves a voicemail and then hangs
up,
Asterisk doesn't know it, exits VoicemailMain2, and loops back to
the
corporate greeting, tying up the line even though the outside
caller
has
hung up.
Therefore, I've added the following hideous hack - er, code - to
voicemail2.c. It starts right after the call to play_and_record()
in
leave_voicemail().
if (res != '#' chan != NULL !strncmp(chan-name, Zap, 3))
{
/* Hang up the Zap channel only */
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}
Obviously, it hangs up the channel after the voicemail has been
recorded,
if the # key wasn't pressed, if the channel still exists, and if
it's
a
Zap channel. I couldn't see a way to do this with AGI.
Question: is this safe? I used a soft hangup because the channel
is
controlled by another thread. I also modified channel.c so that
ast_channel_free() sets chan to NULL after it's freed, just in
case.
Is
there anything else I should be aware of? The code seems to work
in my
testing, resulting in a proper hangup right after the voicemail
has
been
recorded. I'm not up on my Asterisk internals, so I'm not totally
confident about this.
Thanks,
Chris
Re: [Asterisk-Users] Hangups after voicemail
Changing the line signaling is not an option, sorry, since this Asterisk
configuration needs to be flexible and work with any type of analogue line.
Chris
On Wednesday 17 September 2003 09:11, Wade J. Weppler wrote:
Then it sounds like your Telco isn't giving you kewlstart signaling.
This is by far the most reliable method of telling asterisk that the
line has been disconnected. Trying asking your Telco if they can supply
you with Kewlstart or Forward Disconnect Supervision on your line.
Basically, all this does is momentarily reverse the polarity on the line
to indicate that the line has been disconnected. The Zaptel FXO devices
detect this condition to indicate to Asterisk that the line has been
disconnected.
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 12:01 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hangups after voicemail
Hi Wade,
Yes, my zapata.conf looks like this:
group = 1
context = incoming
signalling = fxs_ks
echocancel = yes
echocancelwhenbridged = yes
channel = 1-2
So they are configured as kewlstart.
Thanks,
Chris
On Tuesday 16 September 2003 16:53, Wade J. Weppler wrote:
Have you tried using kewlstart instead? Your loopstart lines might
be
configured for kewlstart (forward disconnect supervision).
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 16, 2003 7:48 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Hangups after voicemail
Hi,
Try as I might, I can't get hangups detected on a Zap channel with
loop
start
lines. So, after someone leaves a voicemail and then hangs up,
Asterisk
doesn't know it, exits VoicemailMain2, and loops back to the
corporate
greeting, tying up the line even though the outside caller has
hung
up.
Therefore, I've added the following hideous hack - er, code - to
voicemail2.c.
It starts right after the call to play_and_record() in
leave_voicemail().
if (res != '#' chan != NULL !strncmp(chan-name, Zap, 3))
{
/* Hang up the Zap channel only */
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}
Obviously, it hangs up the channel after the voicemail has been
recorded,
if
the # key wasn't pressed, if the channel still exists, and if it's
a
Zap
channel. I couldn't see a way to do this with AGI.
Question: is this safe? I used a soft hangup because the channel
is
controlled
by another thread. I also modified channel.c so that
ast_channel_free()
sets
chan to NULL after it's freed, just in case. Is there anything
else I
should
be aware of? The code seems to work in my testing, resulting in a
proper
hangup right after the voicemail has been recorded. I'm not up on
my
Asterisk
internals, so I'm not totally confident about this.
Thanks,
Chris
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Re: [Asterisk-Users] LineJack + Asterisk HELP!
On Wed, 17 Sep 2003 09:06:35 -0300, Bartosz Jozwiak wrote Hello, I have kernel-source-2.4.20.tar.gz and I untar this on. Should I try it once again with tar.bz2 ? I am ranning the same kernel for sure. so where's your kernel source (unpacked)? make a symlink from the directory to /usr/src/linux (ln -sf /usr/src/my-kernel-dir /usr/src/linux) just in case and do a: ls -l /lib/modules/`uname -r`/build make sure it points to your true kernel source. if it doesn't, you're *not* running the kernel you're trying to compile for. if i were you, i'd: * download a fresh vanilla 2.4.22 kernel * untar/bz2 it in /usr/src * make a link from linux-2.4.22 to linux * d/l and install openwall maybe? :) * make menuconfig c. * install the kernel (remember lilo.conf lilo if you use it!) * reboot to the new kernel * do whatever you desire w/the driver hth, grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuration for Asterisk with Cisco Router FXO
Does this FXO gateway have good hangup detection on loop start lines? I configured a Mediatrix gateway (the 1204) similarly to your config, and it didn't detect hangups properly either. Right now, I just have two X100P-type cards in the server (actually, they are cheap generic modem cards and I hacked the wcfxo module to recognise them - saves a bundle) and modified the Asterisk source directly to be smart enough to know when to do a soft hangup. On Wednesday 17 September 2003 09:26, Gerry Boudreaux wrote: Since I see so many questions about this, and could not find a concise answer when I was looking for the same thing... Here is an example showing how to configure communications between Asterisk and a Cisco 2600 router with an FXO card in it. http://www.tape.net/~gerry/asterisk/cisco26x0.html Comments and suggestions are welcome. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Configuration for Asterisk with Cisco Router FXO
I have a Cisco 2600 with a T1 CSU/DSU. Would that work the same way? I would really like to see if I can use the cisco csu/dsu rather than buying a 500 dollar T1 card... On Wed, 17 Sep 2003 11:26:24 -0500, Gerry Boudreaux [EMAIL PROTECTED] wrote: Since I see so many questions about this, and could not find a concise answer when I was looking for the same thing... Here is an example showing how to configure communications between Asterisk and a Cisco 2600 router with an FXO card in it. http://www.tape.net/~gerry/asterisk/cisco26x0.html Comments and suggestions are welcome. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Using M2, Opera's revolutionary e-mail client: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NEW Asterisk Security vulnerability report ...
Hello, There is a new asterisk vulnerability report at this address: http://www.securiteam.com/unixfocus/5HP0H1PB5S.html This is the second security report regarding asterisk for 8 days (http://www.securiteam.com/securitynews/5LP0720B5G.html) Both fixes was reported and fixed silently. My question is: Is it possible in the future such a security problems to be reported in this mailing list or some other security related list? Lubo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hangups after voicemail
You can still use timeouts. The responsetimeout would only be active in
the main menu. If they don't respond within the timeout, then hangup on
them.
The alternative is to loop the menu a set number of times before hanging
up. This would require some logic.
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 12:41 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hangups after voicemail
Yes, I had to set it high, otherwise it didn't work right...I'll
fiddle
with
it a bit.
Timeouts aren't really an option, because if the caller presses #
after
leaving a voicemail then they should be popped back into the main
menu. If
I
could check DTMF signals from extensions.conf, then this would indeed
work.
Basically, the logic after Voicemail2 exits should be
if (last key was not # and the channel is still alive)
hangup
else
goto the main menu
The only way I could do this was by modifying voicemail2.c. So, back
to my
main question: are there any problems with this? I'm most concerned
about
memory issues. Should I be freeing something first, making another
cleanup
function call, etc.?
On Wednesday 17 September 2003 09:08, Wade J. Weppler wrote:
Silencethreshold of 256 sounds a bit high...
You can also add a timeout extension to just hangup the line:
exten = t,1,Hangup
Without using Kewlstart, there isn't anyway for Asterisk to know
that
the line has been disconnected, so you'll have to use the timeouts.
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 11:59 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hangups after voicemail
Hi Martin,
Yes, silence detection in voicemail is working. I am using
Voicemail2
with
the
silencethreshold set to 256. However, the line doesn't hang up
after
the
silence is detected; instead, Voicemail2 exits after recording the
voicemail
correctly, and Asterisk loops back into the main menu as if the #
key
was
pressed because the channel is still alive. Then it times out
after 15
seconds, as you can see below.
From extensions.conf:
[incoming]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,BackGround(corp_greeting)
include = locals
include = errors
The locals context consists of macros which look like this:
exten = s,1,Playback(transfer,skip)
exten = s,2,Dial(${ARG2},20)
exten = s,3,Voicemail2(u${ARG1})
exten = s,4,Goto(incoming,s,1)
exten = s,103,Voicemail2(b${ARG1})
exten = s,104,Goto(incoming,s,1)
So after a voicemail is left, there is a Goto back into the
incoming
context.
It all works great, except for when the line gets tied up by the
DigitTimeout
and ResponseTimeout bits when hangups aren't detected.
I've tried using BUSYDETECT_MARTIN with busydetect=yes and it
didn't
work.
The
channel stays up after the outside caller hangs up.
Since all of our inside phones are SIP lines, there is no problem
detecting
hangups when a voice conversation is taking place, since Asterisk
obviously
detects SIP hangups correctly whether it's SIP to SIP or SIP to
outside
line.
The problem is really only when outside callers leave voicemail.
Thanks,
Chris
On Wednesday 17 September 2003 08:09, Martin Pycko wrote:
Do you have silence in the channel when the remote user hangs up
or
busy
tone ?
If you have silence you can use maxsilence=x_seconds in
voicemail.conf
with
Voicemail2 application and that will make sure the calls are
hanged
up
after x_seconds of silence in the channel.
If you have busy tone then use the busydetect=yes in
zapata.conf.
You can also limit the length of the voicemail message with
maxmessage=x_seconds in the voicemail.conf
regards
Martin
On Tue, 16 Sep 2003, Christian Hecimovic wrote:
Hi,
Try as I might, I can't get hangups detected on a Zap channel
with
loop
start lines. So, after someone leaves a voicemail and then
hangs
up,
Asterisk doesn't know it, exits VoicemailMain2, and loops back
to
the
corporate greeting, tying up the line even though the outside
caller
has
hung up.
Therefore, I've added the following hideous hack - er, code -
to
voicemail2.c. It starts right after the call to
play_and_record()
in
leave_voicemail().
if (res != '#' chan != NULL !strncmp(chan-name, Zap,
3))
{
/* Hang up the Zap channel only */
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}
Obviously, it hangs up the channel after the voicemail has
been
recorded,
if the # key wasn't pressed, if the channel still
Re: [Asterisk-Users] Adpcm, 6KHz codec
Alex Zarubin wrote: I am positive, 4 bits per sample, 6000 Hz. This is a default play/record setting for the older Dialogic R4 API and we need to play zillions (sic!) of files (messages) recorded this way. Conversion issues: - expensive C versions of the OKI/Dialogic ADPCM codec are freely available. - resampling quality It does need resampling, as the codec will give you linear PCM at 6000/s. The Dialogic cards have to do that anyway, to make an 8000/s stream for an A-law or u-law PCM channel. You won't loose more quality than they do (unless you do something dumb). - storage - application changes - etc. Would be real nice and useful to have this codec. Thank you. Alex Zarubin Webley Systems, Inc. Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sample paging config
Hi, Can someone please post a sample config (oss.conf, extensions.conf, etc.) of what is necessary to use the soundcard in the Asterisk server to do overhead paging? Thank you. Travis Microserv ; -- oss.conf ; ; Open Sound System Console Driver Configuration File ; [general] ; ; Automatically answer incoming calls on the console? Choose yes if ; for example you want to use this as an intercom. ; autoanswer=yes ; ; Default context (is overridden with @context syntax) ; context=local ; ; Default extension to call ; extension=s ; ; Default language ; ;language=en ; ; Silence supression can be enabled when sound is over a certain threshold. ; The value for the threshold should probably be between 500 and 2000 or so, ; but your mileage may vary. Use the echo test to evaluate the best setting. ;silencesuppression = yes ;silencethreshold = 1000 ; -- extensions.conf ; exten = 111,1,Dial(CONSOLE/dsp) exten = 111,2,Hangup You will need to have the chan_oss module correctly loaded to use the console as a pager. See show modules to verify that it's been installed correctly. JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sip call waiting
Hi folks, As none of the SIP softphones that I tested can disable more than one incoming call, I decided to implement it by software ;-) I'm attaching a patch that does it. To make it work, modify your sip.conf file and include callwaiting=[0|1] at the general section, or for each peer that you wish to control. Please note that I haven't tested it too much, and my source tree is quite old, so I'm not sure if this patch will apply to the current CVS. Let me know if you find something wrong asap, as this goes into production tomorrow ! Best regards, PauloHM Paulo - Have you tried using the already-existing feature of outgoinglimit= in sip.conf? I have not tried it as a call waiting canceller, but you might be able to set it to 1 to get what you want. http://bugs.digium.com/bug_view_page.php?bug_id=098 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hangups after voicemail
set silencethreshold to 50 and before voicemail call
responsetimeout,0
regards
Martin
On Wed, 17 Sep 2003, Christian Hecimovic wrote:
Hi Martin,
Yes, silence detection in voicemail is working. I am using Voicemail2 with the
silencethreshold set to 256. However, the line doesn't hang up after the
silence is detected; instead, Voicemail2 exits after recording the voicemail
correctly, and Asterisk loops back into the main menu as if the # key was
pressed because the channel is still alive. Then it times out after 15
seconds, as you can see below.
From extensions.conf:
[incoming]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,BackGround(corp_greeting)
include = locals
include = errors
The locals context consists of macros which look like this:
exten = s,1,Playback(transfer,skip)
exten = s,2,Dial(${ARG2},20)
exten = s,3,Voicemail2(u${ARG1})
exten = s,4,Goto(incoming,s,1)
exten = s,103,Voicemail2(b${ARG1})
exten = s,104,Goto(incoming,s,1)
So after a voicemail is left, there is a Goto back into the incoming context.
It all works great, except for when the line gets tied up by the DigitTimeout
and ResponseTimeout bits when hangups aren't detected.
I've tried using BUSYDETECT_MARTIN with busydetect=yes and it didn't work. The
channel stays up after the outside caller hangs up.
Since all of our inside phones are SIP lines, there is no problem detecting
hangups when a voice conversation is taking place, since Asterisk obviously
detects SIP hangups correctly whether it's SIP to SIP or SIP to outside line.
The problem is really only when outside callers leave voicemail.
Thanks,
Chris
On Wednesday 17 September 2003 08:09, Martin Pycko wrote:
Do you have silence in the channel when the remote user hangs up or busy
tone ?
If you have silence you can use maxsilence=x_seconds in voicemail.conf
with
Voicemail2 application and that will make sure the calls are hanged up
after x_seconds of silence in the channel.
If you have busy tone then use the busydetect=yes in zapata.conf.
You can also limit the length of the voicemail message with
maxmessage=x_seconds in the voicemail.conf
regards
Martin
On Tue, 16 Sep 2003, Christian Hecimovic wrote:
Hi,
Try as I might, I can't get hangups detected on a Zap channel with loop
start lines. So, after someone leaves a voicemail and then hangs up,
Asterisk doesn't know it, exits VoicemailMain2, and loops back to the
corporate greeting, tying up the line even though the outside caller has
hung up.
Therefore, I've added the following hideous hack - er, code - to
voicemail2.c. It starts right after the call to play_and_record() in
leave_voicemail().
if (res != '#' chan != NULL !strncmp(chan-name, Zap, 3)) {
/* Hang up the Zap channel only */
ast_softhangup(chan, AST_SOFTHANGUP_EXPLICIT);
}
Obviously, it hangs up the channel after the voicemail has been recorded,
if the # key wasn't pressed, if the channel still exists, and if it's a
Zap channel. I couldn't see a way to do this with AGI.
Question: is this safe? I used a soft hangup because the channel is
controlled by another thread. I also modified channel.c so that
ast_channel_free() sets chan to NULL after it's freed, just in case. Is
there anything else I should be aware of? The code seems to work in my
testing, resulting in a proper hangup right after the voicemail has been
recorded. I'm not up on my Asterisk internals, so I'm not totally
confident about this.
Thanks,
Chris
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Re: [Asterisk-Users] Hangups after voicemail
Hi Wade,
If you scroll down a bit, you'll see my incoming context from extensions.conf.
It has exactly that: a 15 second timeout. I guess I could just shorten this
to, say, 10 seconds or something, but when you only have two lines, tying one
up for any longer than necessary is undesireable.
Basically, I want to beat the loop start hangup detection problem with a bit
of logic. When someone has finished leaving voicemail, say, and they haven't
pressed #, then I'm 99% certain they've hung up. That sort of thing. That's
what I've done with my little code modification.
On Wednesday 17 September 2003 09:58, Wade J. Weppler wrote:
You can still use timeouts. The responsetimeout would only be active in
the main menu. If they don't respond within the timeout, then hangup on
them.
The alternative is to loop the menu a set number of times before hanging
up. This would require some logic.
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 12:41 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hangups after voicemail
Yes, I had to set it high, otherwise it didn't work right...I'll
fiddle
with
it a bit.
Timeouts aren't really an option, because if the caller presses #
after
leaving a voicemail then they should be popped back into the main
menu. If
I
could check DTMF signals from extensions.conf, then this would indeed
work.
Basically, the logic after Voicemail2 exits should be
if (last key was not # and the channel is still alive)
hangup
else
goto the main menu
The only way I could do this was by modifying voicemail2.c. So, back
to my
main question: are there any problems with this? I'm most concerned
about
memory issues. Should I be freeing something first, making another
cleanup
function call, etc.?
On Wednesday 17 September 2003 09:08, Wade J. Weppler wrote:
Silencethreshold of 256 sounds a bit high...
You can also add a timeout extension to just hangup the line:
exten = t,1,Hangup
Without using Kewlstart, there isn't anyway for Asterisk to know
that
the line has been disconnected, so you'll have to use the timeouts.
-wade
-Original Message-
From: Christian Hecimovic [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 17, 2003 11:59 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Hangups after voicemail
Hi Martin,
Yes, silence detection in voicemail is working. I am using
Voicemail2
with
the
silencethreshold set to 256. However, the line doesn't hang up
after
the
silence is detected; instead, Voicemail2 exits after recording the
voicemail
correctly, and Asterisk loops back into the main menu as if the #
key
was
pressed because the channel is still alive. Then it times out
after 15
seconds, as you can see below.
From extensions.conf:
[incoming]
exten = s,1,Answer
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,BackGround(corp_greeting)
include = locals
include = errors
The locals context consists of macros which look like this:
exten = s,1,Playback(transfer,skip)
exten = s,2,Dial(${ARG2},20)
exten = s,3,Voicemail2(u${ARG1})
exten = s,4,Goto(incoming,s,1)
exten = s,103,Voicemail2(b${ARG1})
exten = s,104,Goto(incoming,s,1)
So after a voicemail is left, there is a Goto back into the
incoming
context.
It all works great, except for when the line gets tied up by the
DigitTimeout
and ResponseTimeout bits when hangups aren't detected.
I've tried using BUSYDETECT_MARTIN with busydetect=yes and it
didn't
work.
The
channel stays up after the outside caller hangs up.
Since all of our inside phones are SIP lines, there is no problem
detecting
hangups when a voice conversation is taking place, since Asterisk
obviously
detects SIP hangups correctly whether it's SIP to SIP or SIP to
outside
line.
The problem is really only when outside callers leave voicemail.
Thanks,
Chris
On Wednesday 17 September 2003 08:09, Martin Pycko wrote:
Do you have silence in the channel when the remote user hangs up
or
busy
tone ?
If you have silence you can use maxsilence=x_seconds in
voicemail.conf
with
Voicemail2 application and that will make sure the calls are
hanged
up
after x_seconds of silence in the channel.
If you have busy tone then use the busydetect=yes in
zapata.conf.
You can also limit the length of the voicemail message with
maxmessage=x_seconds in the voicemail.conf
regards
Martin
On Tue, 16 Sep 2003, Christian Hecimovic wrote:
Hi,
Try as I might, I can't get hangups detected on
Re: [Asterisk-Users] Sip call waiting
Have you tried using the already-existing feature of outgoinglimit= in sip.conf? I have not tried it as a call waiting canceller, but you might be able to set it to 1 to get what you want. Have outgoinglimit= and incominglimit= both been commited to CVS? Can these be applied globally and per UA config, or are they set per UA config only? Thanks. -- __ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 PRI
Well, it turns out that I needed to get intimate with the libpri source anyway. After starting * with libpri linked in, * would annouce that the B channels were up, but the Definity saw them as out of service (OOS/FE-PINS initially and then after two minutes, out-of-service-FE). I started comparing the messages exchanged when the DS1 comes up to those on a Dialogic GC box, and ended up patching libpri. The response to SERVICE(0x0f) should be SERVICE ACKNOWLEDGE (0x07), as is indicated in the comments. However, the code that's currently in CVS seems to be modifying a byte in the Call Reference, not the Message Type byte. (Perhaps this has something to do with my build environment?!? The typedef for q931_h looks pretty tame, and I'm building on Debian stable, gcc 2.95.4.) Anyway, here's the patch: [EMAIL PROTECTED]:/usr/src$ diff -u ./libpri/q931.c ./libpri_prev/q931.c --- ./libpri/q931.c Tue Sep 16 14:29:21 2003 +++ ./libpri_prev/q931.cTue Sep 9 16:49:10 2003 @@ -1840,7 +1840,7 @@ /* This is the weird maintenance stuff. We majorly KLUDGE this by changing byte 4 from a 0xf (SERVICE) to a 0x7 (SERVICE ACKNOWLEDGE) */ - h-raw[4] -= 0x8; + h-raw[3] -= 0x8; q931_xmit(pri, h, len, 1); return 0; } After recompiling libpri, I'm up and running. Being new to the list and the project, I'd appreciate some feedback on whether or not this sort of thing is appropriate for submission back into CVS. For one, I'm not sure if the maintenance kludge is Definity-specific. (/me probably needs to take a look at asterisk-dev list.) Cheers, tony ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip call waiting
Damn. Seems to implement what I was looking for ... ;-( Does anyone know if the incominglimit works if the call is being generated from a queue? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: September 17, 2003 2:19 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Sip call waiting Hi folks, As none of the SIP softphones that I tested can disable more than one incoming call, I decided to implement it by software ;-) I'm attaching a patch that does it. To make it work, modify your sip.conf file and include callwaiting=[0|1] at the general section, or for each peer that you wish to control. Please note that I haven't tested it too much, and my source tree is quite old, so I'm not sure if this patch will apply to the current CVS. Let me know if you find something wrong asap, as this goes into production tomorrow ! Best regards, PauloHM Paulo - Have you tried using the already-existing feature of outgoinglimit= in sip.conf? I have not tried it as a call waiting canceller, but you might be able to set it to 1 to get what you want. http://bugs.digium.com/bug_view_page.php?bug_id=098 JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Follow Me
At 06:48 PM 9/16/2003, you wrote:
cell phone into the call (or my office number, etc.) I understand the
selected numbers part of it, but not how to get it to use the three way. If
I send it to Nufone first, I'm paying for a call to a local number (my
cell) that I don't need to.
This should work...
[default]
exten = s,1,Dial(Zap/3,20,t) ; This is your desk phone
exten = s,2,Dial(Zap/2/1234567,20,t) ; This is your secondary POTS line
calling your office
exten = s,3,Dial(Zap/2/3217654,20,t) ; This is your secondary POTS line
calling your cell phone
; I've never tried this one coming up, but I think it's worth a shot as it
works just fine for local extensions
exten = s,4,Dial(Zap/2/3217654Zap/3/3217654,20,t) ; This is your
secondary and tertiary POTS lines calling your cell phone anbd office
As long as none of these lines go to voicemail, they should fail over
properly in order. You can also make it more complicated with time-based
includes and gotos.
--Ernest
At 09:57 AM 9/16/2003 -0700, Ernest W. Lessenger wrote:
At 11:22 PM 9/14/2003, you wrote:
First -- Thanks to everyone who offered their help and tips on getting my
Cisco 7960 working with Asterisk -- this is great stuff.
Does anyone have any examples of Follow Me or other call forwarding with
a single PSTN interface? Or a pointer on what I need to read to figure it
out?
Is this what you need? Basically, the local trunk and the Nufone trunk
fail over to each other. So, if you have a forward set up and transfer to
a non-local extension, the call will go out even if the original incoming
call was made on the PSTN line.
[trunklocal]
exten = _NXX,1,Dial(${TRUNK}/${EXTEN})
exten = _NXX,102,Dial(${NUFONE}/1${AREACODE}${EXTEN})
exten = _NXX,203,Congestion()
[iaxprovider]
exten = _1NXXNXX,1,Dial(${NUFONE}/${EXTEN})
exten = _1NXXNXX,102,Dial(${TRUNK})
exten = _1NXXNXX,203,Congestion()
exten = _011.,1,Dial(${NUFONE}/${EXTEN})
exten = _011.,102,Congestion()
exten = _1011.,1,Dial(${NUFONE}/${EXTEN})
exten = _1011.,102,Congestion()
--Ernest
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[Asterisk-Users] CODECS and thier practical usage stats
Hi, What are real life bandwith stats for * supported codecs? Is it true one can run 6-32 conversations over DSL, as stated in this list? Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming 976 numbers from dialing out.
Just as simple to call your telco and have those turned off then its not an issue ever! bkw On Wed, 17 Sep 2003, Ariel Batista wrote: I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten = 1900XXX,1,Congestion exten = XXX976,1,Congestion exten = XXX976,1,Congestion exten = 1XXX976,1,Congestion exten = 91900XXX,1,Congestion exten = 9XXX976,1,Congestion exten = 91XXX976,1,Congestion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CODECS and thier practical usage stats
Thats all going to depend on the speed of your DSL... bkw On Wed, 17 Sep 2003, Senad Jordanovic wrote: Hi, What are real life bandwith stats for * supported codecs? Is it true one can run 6-32 conversations over DSL, as stated in this list? Senad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Programming 976 numbers from dialing out.
I think that you should put a _ at the beggining of each string to show that it is a pattern to be matched instead of a literal extension. Sean ___ Sean Robertson NETXUSA p. 800-289-6389 f. 864-233-4344 Ask me about Voice over IP. http://www.netxusa.com/ - Original Message - From: Ariel Batista [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 17, 2003 4:36 PM Subject: [Asterisk-Users] Programming 976 numbers from dialing out. I would like to prevent * from dialing 900 and 976 numbers. I setup the following settings in extensions.conf. But this does not seem to work! I don't know what I am doing wrong please help! exten = 1900XXX,1,Congestion exten = XXX976,1,Congestion exten = XXX976,1,Congestion exten = 1XXX976,1,Congestion exten = 91900XXX,1,Congestion exten = 9XXX976,1,Congestion exten = 91XXX976,1,Congestion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm - IAX client for Win32
On Wed, 17 Sep 2003 15:14:27 -0500, Josh Roberson [EMAIL PROTECTED] wrote: The copy I downloaded from the website never did register with *. It would make authenticated calls, but wouldn't actually register with the server. Even checked the IAX peers, and nope, wasn't registered. Do you see anything with iax debug? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Van Donselaar Sent: Wednesday, September 17, 2003 1:00 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxComm - IAX client for Win32 On Wed, 17 Sep 2003 11:27:25 +0200, Florian Overkamp [EMAIL PROTECTED] wrote: At 19:55 16-9-2003 -0500, you wrote: iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome. Well, this looks like a big improvement, but I cant seem to find the option to register at the asterisk server. Is it impossible, or am I missing it ? Would be a hefty requirement for real use, I think... It automatically registers with all asterisk servers that have been configured in the Options|Directory dialog. I dial out and register from two different servers. I previously had an auto register checkbox, but changed to registering all servers when I moved the servers list from a listcontrol to a combobox. I'm thinking that you would want to register with any server through which you may want to make outbound calls. When the servers are read from the registry, they are read in alphabetical order, and registration is attempted in that order. (The order may be different on other platforms). You should see Registration accepted in the status bar after the last server is registered. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.515 / Virus Database: 313 - Release Date: 9/1/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Programming 976 numbers from dialing out.
You need to add a _ at the start of the string, to trigger pattern matching. Eg: exten = _1900XXX,1,Congestion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Web Based Management App
Hi All Appologies if this has been asked before, or if this is not the correct place to ask just point me in the right direction. Is there a web based management application available for asterisk?? If not is there any interest in developing one? I am good at writing in PHP and don't think it would be too difficult to put something together. Thanks Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI AVM Fritz DID question
Hi, Does anyone know if the CAPI driver and the AVM Fritz card support DID in the same way as the zaptel E1 interface.e.g [incoming] exten = 2054286275,1,Goto,default|6275|1 exten = _2054286XXX,1,Goto,hsvorlss|s|1 etc etc Regards Adrian Brown --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.518 / Virus Database: 316 - Release Date: 11/09/2003 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and ACD system
Hello List, I would like you to help me on solving our problem. Our possible deployment is as follows: PSTN --- asterisk -- PBX ACD system phone set I would like to know is there any way to let asterisk know which agent picks the phone call via the ACD system. can it be obtained via PBX CDR or signalling between pbx and asterisk or asterisk can know from the call set up ?? Regards, George Lin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem loading chan_iax2.so and chan_zap.sofrom latest CVS
Steven Thanks for the help. After rebooting the box, * gives me an error claiming that * is already running on /var/run/asterisk.ctl Before I rebooted I ensured that there was no asterisk.pid or asterisk.ctl. After I get the above mentioned message if I then run asterisk again I get the Unable to open...device busy (with or without the asterisk.pid and/or asterisk.ctl) Any help would be greatly appreciated. Rgds Dan - Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, September 17, 2003 12:02 AM Subject: Re: [Asterisk-Users] problem loading chan_iax2.so and chan_zap.sofrom latest CVS On Tue, 2003-09-16 at 20:27, Dan Fernandez wrote: I just updated to the new CVS and now I am getting the following error from chan_zap (modprobe wcfxo works fine): WARNING[16384]: File chan_zap.c, Line 577 (zt_open): Unable to specify channel 1: Device or resource busy snip WARNING:Unable to bind to 0.0.0.0 port 4569: Address already in use This looks rather obvious to me that you may not have stopped the previous asterisk install. Either that or you have a kernel problem and (oddly) need to reboot to free the port and the device handles. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and ACD system
Do you mean that you're going to take trunks in to Asterisk, then feed extensions out which are connected to trunk ports on a traditional PBX/ACD system? If that's the case, there may be some info in CDRs but generally it wouldn't be until after the call was complete, not at the point that the agent answers the call. Help any? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ITFS VoIP
I'm looking for toll-free #'s in: Germany Australia United Kingdom China Russia Singapore Netherlands that ring to a US based PSTN #. I've contacted people like QWest, XO, etc.. and their rates are extremely high ($1.74/min from the UK). Is there a better way to do this that involves VoIP? Thanks, Justin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Analog FXO Card
If you really want to save some money cut Digium out of their well deserved $$$, you can find this same device for less than $10 - you'll need to put your own heat sink on. John Ternovas wrote: If anyone is looking, I just ran accross an ebay auction for X100P Cards at what I thought was a very reasonable price. http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3046843672category=48483rd=1 http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3046843672category=48483rd=1 Do you Yahoo!? Yahoo! SiteBuilder http://us.rd.yahoo.com/evt=10469/*http://sitebuilder.yahoo.com - Free, easy-to-use web site design software ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem loading chan_iax2.so and chan_zap.sofrom latest CVS
On Wed, 2003-09-17 at 17:10, Dan Fernandez wrote: Steven Thanks for the help. After rebooting the box, * gives me an error claiming that * is already running on /var/run/asterisk.ctl Before I rebooted I ensured that there was no asterisk.pid or asterisk.ctl. After I get the above mentioned message if I then run asterisk again I get the Unable to open...device busy (with or without the asterisk.pid and/or asterisk.ctl) Any help would be greatly appreciated. First, please do not group reply/reply all to this lists mail. I will get a copy nearly as fast through the list as I will to my own mailbox. Since I have all mail filtered to specific content based mailboxes your personal copy to me drops in my INBOX and not in the appropriate folder like the copy from the list does. I know some lists need you to do this because they do not set the reply-to address back to the list. This is one of the saner lists that does the right think in my opinion. Now to your problem. What command line are you using to launch asterisk? Sounds like you are using something like asterisk -vvv Try adding a c to the end so you get a command line at the end of start up. or just issue a asterisk -r to connect to the currently running copy. The device busy message is telling you a application is already running with access to those files. issue a ps -axuwww|grep [a]sterisk and see if it shows up. If you see any lines show up, it is already running and the asterisk -r will connect you to it. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ITFS VoIP
Not sure about VoIP, but check the URL for termination of UK 0800 numbers in to the USA. http://www.freephonenumbers.co.uk/numbers/international.htm Cheers Paul ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iconnecthere Problem
I can't seem to dial out with Iconnecthere. I am using the following commands. I get a message that the session is in progress and the call never goes through. Can anyone confirm if iconnect is working and if I am missing something? Thanks, Kevin in sip.conf: [iconnect] type=friend insecure=yes port=5060 username=xyz secret=abc host=natrelay.deltathree.com dtmfmode=inband callerid=15408675512 nat=yes in extensions.conf: exten = 8500,1,Dial(SIP/[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] documentation?
-= On Wed, 17 Sep 2003 11:01:34 -0600, Rich Adamson [EMAIL PROTECTED] said: Examples, Where should I have learned that *8# is the call pickup dialing sequence? A good question. I didn't know about any of them until James Sizemore posted this handy list on Sept.8: *0# sends flash *8# remote call pickup (pickup phone in your group) *67# disable caller id *70# no call waiting *78# do not disturb on *79# do not disturb off *72# enable call forwarding *73# disable call forwarding *82# enable callerid All news to me. :) I do a lot of google searching on the Asterisk archives: http://www.google.com/custom?sitesearch=lists.digium.com Where should I have learned about the t and r in: exten = s,1,Dial(SIP/3000,20,tr) and all the other possible options in various * config statements? The first thing I try is show application Dial or somesuch. Then go grep for the application in the sample config directory to see every instance of how it's used in an example. Then look at the source. It took me an embarrassingly long time to figure out the LEADING arguments to Goto() are optional.. it's a bit backwards: Goto(context,extension,priority) Goto(extension,priority) Goto(priority) It wasn't my first guess, so my menus were all wacky. There are some inconsistencies that will probably work themselves out over time, like the whole: App,arg1,arg2 App(arg1|arg2) App(arg1,arg2) I can't quite figure out if some things still *require* the vertical pipe, like going to another extensions: 400 = Goto(139343234|1) All in all, the best resources have been this list and the handbook. Working examples are the best.. a 'cookbook' would be awesome. -Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Iconnecthere Problem
I got it working by preceding the number with . However, the quality isn't very good and I have problems transmitting DTMF reliably. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 17, 2003 6:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Iconnecthere Problem I can't seem to dial out with Iconnecthere. I am using the following commands. I get a message that the session is in progress and the call never goes through. Can anyone confirm if iconnect is working and if I am missing something? Thanks, Kevin in sip.conf: [iconnect] type=friend insecure=yes port=5060 username=xyz secret=abc host=natrelay.deltathree.com dtmfmode=inband callerid=15408675512 nat=yes in extensions.conf: exten = 8500,1,Dial(SIP/[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [Release] Skinny Support in cvs
So I've been trying to pay attention, but I hadn't seen any updates on SourceForge. I inferred from the thread I could get a copy using CVS, but it looks like our firewall is keeping me out of CVS. Is there another way to come by the source? Dan -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Saturday, September 13, 2003 9:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] [Release] Skinny Support in cvs If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s). I have personally tested this code on 7910 and 12SP+'s and will soon dive into a 7960. There currently may be issues with 7920s and ATAs, but with some proper debug information and/or the acutal device in my grubby mitts I am sure I can get around any nuances. If you have an issue with this code please use http://bugs.digium.com. Patches are absolutely apprecaited, however you should check with me before spending time as it may be a feature I have already played with locally and haven't gotten around to intergrating it into the mainline CVS code. I would like to thank miro_ for his patience and fnancial support, along with [Sim], klasstek, bkw_, PavelL, theo and ManxPower for willingly diving into nearly untested code and debuging. Lastly, we cannot forget Mark Spencer for this absolutely amazing piece of software! A quick sample config: skinny.conf: ; Typical config for a 7910 [jeremy] ; Device name device=SEP0007EB363201 ; Offical identifier (SEP+mac adress) context=default line = 500 extensions.conf: exten = 1234,1,Dial,SKINNY/[EMAIL PROTECTED]|25|r Disclaimer: All research and development of chan_skinny is for the sole purpose of writing interoperable software under Sect. 1201 (f) Reverse Engineering exception of the DMCA. The Skinny Client Control Protocol is a Cisco Systems Incorporated Trademark. chan_skinny is distributed WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [Release] Skinny Support in cvs
Ya fire your network admin :P Firewalls shouldn't be blocking cvs and such if they do then your admin is way too anal. bkw On Wed, 17 Sep 2003, Dan Austin wrote: So I've been trying to pay attention, but I hadn't seen any updates on SourceForge. I inferred from the thread I could get a copy using CVS, but it looks like our firewall is keeping me out of CVS. Is there another way to come by the source? Dan -Original Message- From: Jeremy McNamara [mailto:[EMAIL PROTECTED] Sent: Saturday, September 13, 2003 9:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] [Release] Skinny Support in cvs If you have been paying attention, you already know this, but this weekend I have spent time ironing out the various details with my chan_skinny code that has been out there, if you knew where to look. I believe I now have all basic features operational and am going to be working on getting the class 5 (hold, transfers, call waiting and caller*id, etc) operational in the comming week(s). I have personally tested this code on 7910 and 12SP+'s and will soon dive into a 7960. There currently may be issues with 7920s and ATAs, but with some proper debug information and/or the acutal device in my grubby mitts I am sure I can get around any nuances. If you have an issue with this code please use http://bugs.digium.com. Patches are absolutely apprecaited, however you should check with me before spending time as it may be a feature I have already played with locally and haven't gotten around to intergrating it into the mainline CVS code. I would like to thank miro_ for his patience and fnancial support, along with [Sim], klasstek, bkw_, PavelL, theo and ManxPower for willingly diving into nearly untested code and debuging. Lastly, we cannot forget Mark Spencer for this absolutely amazing piece of software! A quick sample config: skinny.conf: ; Typical config for a 7910 [jeremy] ; Device name device=SEP0007EB363201 ; Offical identifier (SEP+mac adress) context=default line = 500 extensions.conf: exten = 1234,1,Dial,SKINNY/[EMAIL PROTECTED]|25|r Disclaimer: All research and development of chan_skinny is for the sole purpose of writing interoperable software under Sect. 1201 (f) Reverse Engineering exception of the DMCA. The Skinny Client Control Protocol is a Cisco Systems Incorporated Trademark. chan_skinny is distributed WITHOUT ANY WARRANTY; without even the implied warranty of MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Iconnecthere Problem
Try host=sipauth.deltathree.com -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, September 17, 2003 6:46 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Iconnecthere Problem I can't seem to dial out with Iconnecthere. I am using the following commands. I get a message that the session is in progress and the call never goes through. Can anyone confirm if iconnect is working and if I am missing something? Thanks, Kevin in sip.conf: [iconnect] type=friend insecure=yes port=5060 username=xyz secret=abc host=natrelay.deltathree.com dtmfmode=inband callerid=15408675512 nat=yes in extensions.conf: exten = 8500,1,Dial(SIP/[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxComm - IAX client for Win32
If possible, I'd like to get the source code (don't need Linux or Mac) for Windows, please. Also, which C++ compiler should I be using to compile. I have had success with the DOS/prompt version. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Florian Overkamp Sent: Wednesday, September 17, 2003 5:27 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxComm - IAX client for Win32 At 19:55 16-9-2003 -0500, you wrote: iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome. Well, this looks like a big improvement, but I cant seem to find the option to register at the asterisk server. Is it impossible, or am I missing it ? Would be a hefty requirement for real use, I think... Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web Based Management App
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mark Evans wrote: | Hi All | | Appologies if this has been asked before, or if this is not the correct | place to ask just point me in the right direction. | | Is there a web based management application available for asterisk?? | | If not is there any interest in developing one? I am good at writing in | PHP and don't think it would be too difficult to put something together. Hi Mark, p0lar (I'm sorry, I don't know his real name) has written a nice PHP frontend for configuring the * .conf files. It is available in the CVS and is called phpconfig if I remember correctly. I would probably start with that and help expand upon it. Currently, other than that, there is no sort of GUI interface for Asterisk. If you were to write one, I'm sure it would be greatly appreciated :) Thanks, Leif Madsen. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/aPv36gq3eQ0gpNURAmgJAJ9E+hBetHiBksDQocOCgDufg0XV0ACgtcjl l86Dp+UwcDR6/0GorVKEX1I= =N6X9 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxComm - IAX client for Win32
On Wed, 17 Sep 2003 21:01:02 -0400, Uriel Carrasquilla [EMAIL PROTECTED] wrote: If possible, I'd like to get the source code (don't need Linux or Mac) for Windows, please. http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz gets the source code. There are instructions in iaxclient/simpleclient/wx/README on how to instal/prepare mingw and wxwindows. Also, which C++ compiler should I be using to compile. I have had success with the DOS/prompt version. I used mingw, but I think you ought to be able to use Borland if you tweak the makefile. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Florian Overkamp Sent: Wednesday, September 17, 2003 5:27 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxComm - IAX client for Win32 At 19:55 16-9-2003 -0500, you wrote: iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome. Well, this looks like a big improvement, but I cant seem to find the option to register at the asterisk server. Is it impossible, or am I missing it ? Would be a hefty requirement for real use, I think... Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxComm - IAX client for Win32
Thanks a lot. mingw is my cup of tea. Regards, Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Van Donselaar Sent: Wednesday, September 17, 2003 8:32 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxComm - IAX client for Win32 On Wed, 17 Sep 2003 21:01:02 -0400, Uriel Carrasquilla [EMAIL PROTECTED] wrote: If possible, I'd like to get the source code (don't need Linux or Mac) for Windows, please. http://iaxclient.sourceforge.net/snapshots/iaxclient.tar.gz gets the source code. There are instructions in iaxclient/simpleclient/wx/README on how to instal/prepare mingw and wxwindows. Also, which C++ compiler should I be using to compile. I have had success with the DOS/prompt version. I used mingw, but I think you ought to be able to use Borland if you tweak the makefile. Uriel -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Florian Overkamp Sent: Wednesday, September 17, 2003 5:27 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] iaxComm - IAX client for Win32 At 19:55 16-9-2003 -0500, you wrote: iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome. Well, this looks like a big improvement, but I cant seem to find the option to register at the asterisk server. Is it impossible, or am I missing it ? Would be a hefty requirement for real use, I think... Met vriendelijke groet, Florian Overkamp ObSimRef BV (http://www.obsimref.com/) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Prices for new channel banks, patch panels, cables etc.. etc..
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi All, I'm having a tough time trying to find prices from dealers in Canada for some equipment. I am trying to implement an Asterisk box into a small business using 24 FXS ports and 8 FXO ports. I need to find the pricing for all the relevent equipment: cables, patch panels, channel bank chassis, cards etc..etc.. I think I'm going to tie the channel bank into the Asterisk box using a pair of T1's using a pair of digium T400P's. So I'm asking anyone to reply to me off list with either people I can contact to get prices from, websites which list all the equipment, or if you sell this type of hardware, please email me. I have been interested in the CarrierAccess Adit 600 and perhaps the Mediatrix hardware. If someone has a suggested hardware configuration for what I am doing, it would be appreciated!!! Thanks in advance, Leif Madsen. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/aQGD6gq3eQ0gpNURAsGPAJ43RJ1ImZAA9pWFY0cspnPyAE6PcACgulW6 +aUYtAIidSIcAqG4eEriKl4= =ZYfp -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Prices for new channel banks, patch panels, cables etc.. etc..
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Leif Madsen wrote: | I think I'm going to tie the channel bank into the Asterisk box using a | pair of T1's using a pair of digium T400P's. Oops.. I meant pair of T100P's :) Thanks, Leif Madsen. -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.2 (Cygwin) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQE/aQbm6gq3eQ0gpNURAsbzAKDPNCTkYN5zLO254ZERwJbHrLQKZQCfRsMB 4JJBoLQrbFnbG0lSthPKnxo= =DD3v -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 + 5.x Firmware + *
Yes. 30 phones in production environment. No problems so far. :) Travis At 08:21 PM 9/17/2003 -0500, you wrote: Anyone running the 5.x firmware on their 7960's with asterisk? bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] NEW Asterisk Security vulnerability report ...
On Wednesday 17 September 2003 08:15, Lubomir Christov wrote: Hello, There is a new asterisk vulnerability report at this address: http://www.securiteam.com/unixfocus/5HP0H1PB5S.html They lie. My email address is at the top of the cdr_mysql.c source file, and yet I was never contacted. Both fixes was reported and fixed silently. My question is: Is it possible in the future such a security problems to be reported in this mailing list or some other security related list? Sure, why don't you ask the security researchers to post the problem to the -dev list, instead of only on their website (where we get to find out only long after the fact)? -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nufone 800 numbers working?
Is anyone else having trouble dialing 800 numbers through Nufone? I'm getting the SIT tones no matter what number I dial. Normal long distance works fine. I don't think it's my dial plan (it was working previously). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Web Based Management App
Hi Mark, A switchboard type php application would be great, one that could show all current calls / extensions etc and their states? Just my $0.02 - Original Message - From: Mark Evans [EMAIL PROTECTED] To: Asterisk [EMAIL PROTECTED] Sent: Thursday, September 18, 2003 9:04 AM Subject: [Asterisk-Users] Web Based Management App Hi All Appologies if this has been asked before, or if this is not the correct place to ask just point me in the right direction. Is there a web based management application available for asterisk?? If not is there any interest in developing one? I am good at writing in PHP and don't think it would be too difficult to put something together. Thanks Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone 800 numbers working?
Nope works fine here... NUFONE ROCKS! bkw On Wed, 17 Sep 2003, Peter Pauly wrote: Is anyone else having trouble dialing 800 numbers through Nufone? I'm getting the SIT tones no matter what number I dial. Normal long distance works fine. I don't think it's my dial plan (it was working previously). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nufone 800 numbers working?
On Wed, 17 Sep 2003, Brian West wrote: Nope works fine here... NUFONE ROCKS! bkw On Wed, 17 Sep 2003, Peter Pauly wrote: Is anyone else having trouble dialing 800 numbers through Nufone? I'm getting the SIT tones no matter what number I dial. Normal long distance works fine. I don't think it's my dial plan (it was working previously). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
