[asterisk-users] bad sound quality after Redirect
Hi! I'm building an application which allows to dial via the Asterisk Manager Interface using the originate command. There should be an optional conferencing feature. The manager commands are basically: - action: login username: sdjklgdsjg secret: xxx events: on action: originate callerid: 3847438609 priority: 1 exten: 4068439865 async: 1 context: out channel: SIP/sip-gate/0394839405 - Then talk to each other for a while... - action: redirect priority: 1 exten: 1234 context: conference channel: SIP/sip-gate-0868b000 extrachannel: SIP/sip-gate-086a5000 action: logoff - This approach works but results in a bad sound quality after the redirect. The sound seems to be scrambled. Before redirecting the sound quality is quite well, of course. All extensions are called via SIP with the same codec, so no transcoding should occur. The application used for the conference room is AppConference from http://sourceforge.net/projects/appconference/. But even with a simple destination application (e. g. PlayTones or Playback) the sound quality is as bad as with AppConference. So it doesn't seem to be a problem with AppConference itself. The bad sound quality arises only if the ExtraChannel parameter is given to Redirect. Without ExtraChannel the sound quality is still fine. But the second channel is hungup then of course, which is not intended. Has anyone any ideas how to solve this problem? :-) Best regards Franz ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Asterisknow with video and X-Lite not quite working
Hi Dave, yes, Audio is fine, but no video. And as far as I can see, X-Lite (running 3.0 build 34025 here, all Clients have exactly the same version) supports only the h.263 and h.263p video codecs. But I am not quite sure if I enabled these codecs properly. For *now, I have put the allow-lines into the users.conf, for instance, heres my setup (I cencored out email and secret): [6510] fullname = Benedikt Franz secret = ... email = ... cid_number = 6510 zapchan = context = numberplan-custom-1 hasvoicemail = yes hasdirectory = yes hassip = yes hasiax = yes hasmanager = yes callwaiting = yes threewaycalling = yes mailbox = 6510 hasagent = no group = host = dynamic registersip = yes registeriax = yes allow = h263 allow = h263p canreinvite = yes I am not sure if that is a NAT problem, since all users are either on the local area network, or connected through VPN (I have not tested video with those yet, though), however, I will try that. -- Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden: http://www.gmx.net/de/go/browser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [asterisk-users] Re: Asterisknow with video and X-Lite not quiteworking
Yep, I have added these lines to the sip.conf (but I see that I now have the lines to allow the video codecs in both the sip.conf and in the extionsions.conf. I suppose that is not right?): ; Added video support videosupport=yes allow=h261 allow=h263 allow=h263p BTW: How do I properly post a message on this board? If I send an email to asterisk-users@lists.digium.com, only about one out of five attempts succeed, and the message actually appears in the list. Original-Nachricht Datum: Wed, 14 Mar 2007 11:32:22 +0300 Von: Biju [EMAIL PROTECTED] An: \'Asterisk Users Mailing List - Non-Commercial Discussion\' asterisk-users@lists.digium.com CC: Betreff: RE: [asterisk-users] Re: Asterisknow with video and X-Lite not quiteworking Have you added this line in your sip.onf ? videosupport=yes Biju -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benedikt Franz Sent: Wednesday, March 14, 2007 10:14 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: Asterisknow with video and X-Lite not quiteworking Hi Dave, yes, Audio is fine, but no video. And as far as I can see, X-Lite (running 3.0 build 34025 here, all Clients have exactly the same version) supports only the h.263 and h.263p video codecs. But I am not quite sure if I enabled these codecs properly. For *now, I have put the allow-lines into the users.conf, for instance, heres my setup (I cencored out email and secret): [6510] fullname = Benedikt Franz secret = ... email = ... cid_number = 6510 zapchan = context = numberplan-custom-1 hasvoicemail = yes hasdirectory = yes hassip = yes hasiax = yes hasmanager = yes callwaiting = yes threewaycalling = yes mailbox = 6510 hasagent = no group = host = dynamic registersip = yes registeriax = yes allow = h263 allow = h263p canreinvite = yes I am not sure if that is a NAT problem, since all users are either on the local area network, or connected through VPN (I have not tested video with those yet, though), however, I will try that. -- Ist Ihr Browser Vista-kompatibel? Jetzt die neuesten Browser-Versionen downloaden: http://www.gmx.net/de/go/browser ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Feel free - 10 GB Mailbox, 100 FreeSMS/Monat ... Jetzt GMX TopMail testen: www.gmx.net/de/go/mailfooter/topmail-out ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisknow with video and X-Lite not quite working
I do not think that this is a specifically *now related issue, but I would also welcome such a mailing list. Regards Original-Nachricht Datum: Wed, 14 Mar 2007 10:22:34 -0500 Von: Pari Nannapaneni [EMAIL PROTECTED] An: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com CC: Betreff: Re: [asterisk-users] Asterisknow with video and X-Lite not quite working I request every one to post AsteriskNOW specific questions on the asteriskNOW forums - http://forums.digium.com/ I will talk to our administrator to see if i can get a seperate mailing list created for AsteriskNOW. thanks Pari Pari Nannapaneni GUI Developer Digium Inc. -- Feel free - 5 GB Mailbox, 50 FreeSMS/Monat ... Jetzt GMX ProMail testen: www.gmx.net/de/go/mailfooter/promail-out ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisknow with video and X-Lite not quite working
Hello everyone, I have previously asked this question on the asterisk-video list, but I got directed here. I have a setup consisting of asterisknow beta4 (not sure if that is crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the local network. My computer has a USB-Camera installed, and now I would like to do some video calling with it, at least, so that the other user can see me. When I make a call and then click 'Start' (sending video) in the X-Lite client, nothing seems to happen on the other side, but here it says that a video transmission has begun. According to 'sip show codecs', both the h.263 and h.263p codec are supported, and those are also set on either X-Lite clients. I have enabled 'canreinvite' for both users as well, but still the other user can not see me. I can, however, see the cameras view on my computer, so that seems all properly set up. Could anyone help me sort this out? Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P with FXS module problem
Hi list I conncte my Panasonic KX-TCD705TW cordless phone to TDM400P FXS module. When the system boots or reboots, the LED on the backlit of TDM400P OFTEN gets off and dmesg shows problem with FXS module, as follows Zapata Telephony Interface Registered on major 196 Zaptel Version: 1.4.0 Echo Canceller: MG2 ACPI: PCI interrupt :02:08.0[A] - GSI 19 (level, low) - IRQ 177 Freshmaker version: 71 Freshmaker passed register test Timeout waiting for calibration of module 0 Timeout waiting for calibration of module 0 Proslic Failed on Second Attempt to Auto Calibrate Proslic Failed on Second Attempt to Calibrate Manually. (Try -DNO_CALIBRATION in Makefile) Module 0: FAILED FXS (FCC) Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Installed -- AUTO FXO (FCC mode) Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules) Registered tone zone 14 (Taiwan) When this occurs, no dialtone is heard after off-hook. I have to rmmod wctdm, unplug the phone cable from TDM400P, modprobe wctdm, and plug phone cable to make this line work. The point is ztcfg without rmmod wctdm cannot make working. I run asterisk 1.4.0 / zaptel 1.4.0 on Debian 3.1r4. I compile from source. Any help or recommendation will be appreciated. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] background() with m option
I have same problem and no mailing list response. I suggest we go for
reporting bug.
- Original Message -
From: Jack Wei [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, January 26, 2007 1:16 AM
Subject: [asterisk-users] background() with m option
Hi...
In my dialplan, I have the following:
exten = s,1,Background(${RECORDING}|m)
exten = s,n,Voicemail(${DID_NO})
exten = 0,1,Voicemail(${DID_NO})
exten = a,1,VoiceMailMain(${DID_NO})
exten = h,1,Hangup
In version 1.2, when I hit 0 during the playback, I will be directed to
voicemail. But in verison 1.4, the call hangs up.
[Jan 24 16:05:37] DTMF[5754]: channel.c:2148 __ast_read: DTMF begin '0'
received on SIP/5060-08c53e68
[Jan 24 16:05:37] DTMF[5754]: channel.c:2128 __ast_read: DTMF end '0'
received on SIP/5060-08c53e68
== Spawn extension (play_recording, s, 1) exited non-zero on
'SIP/5060-08c53e68'
-- Executing [EMAIL PROTECTED]:1] Hangup(SIP/5060-08c53e68, ) in new
stack
== Spawn extension (play_recording, h, 1) exited non-zero on
'SIP/5060-08c53e68'
Does anyone tell me why this is happening?
Thanks,
Jack
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[asterisk-users] cmd Backgound problem with option m
Hi list
I encountered problem in using Background command. Below is my
extensions.conf.
[mainmenu]
exten = 4,1,Wait(1)
exten = 4,2,Background(thank-you-for-calling)
exten = 4,3,Goto(n01|s|1)
[n01]
exten = s,1,NoOp(${CONTEXT})
exten = s,2,Background(thank-you-cooperation|m)
exten = s,3,WaitExten()
exten = s,4,Playback(digits/pound)
exten = 1,1,Playback(digits/1)
exten = i,1,Playback(digits/star)
Without m option, everything's fine.
If m option is present and when sound is playing,
- pressing 1 terminates the call and does not goto ext 1
- pressing any other key does not stop sound playing, as expected.
the message on the manager interface when 1 pressing.
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-for-calling)
in new stack
-- Playing 'thank-you-for-calling' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, n01|s|1) in new stack
-- Goto (n01,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, n01) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1,
thank-you-cooperation|m) in
new stack
-- Playing 'thank-you-cooperation' (language 'en')
== Spawn extension (n01, s, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
I also tried Background(thank-you-cooperation|m||n01). The result is
- pressing 1 goto ext i
- pressing any other key does not stop sound playing, as expected.
the message on the manager interface when 1 pressing.
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-cooperation)
in new stack
-- Playing 'thank-you-cooperation' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, n01|s|1) in new stack
-- Goto (n01,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, n01) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1,
thank-you-cooperation|m||n01)
in new stack
-- Playing 'thank-you-cooperation' (language '')
-- Sent into invalid extension 'E8' in context 'n01' on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] Playback(Zap/1-1, digits/star) in new
stack
-- Playing 'digits/star' (language 'en')
== Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'
NOTICE that * tries to go to ext 'E8' which is a French alphabet e with
grave accent.
DTMF detection problem? but if context option and m option of Background is
not specified, everything works well.
any help will be appreciated.
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[asterisk-users] cmd Backgound problem with option m
Hi list
I encountered problem in using Background command. Below is my
extensions.conf.
[mainmenu]
exten = 4,1,Wait(1)
exten = 4,2,Background(thank-you-for-calling)
exten = 4,3,Goto(n01|s|1)
[n01]
exten = s,1,NoOp(${CONTEXT})
exten = s,2,Background(thank-you-cooperation|m)
exten = s,3,WaitExten()
exten = s,4,Playback(digits/pound)
exten = 1,1,Playback(digits/1)
exten = i,1,Playback(digits/star)
Without m option, everything's fine.
If m option is present and when sound is playing,
- pressing 1 terminates the call and does not goto ext 1
- pressing any other key does not stop sound playing, as expected.
the message on the manager interface when 1 pressing.
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-for-calling)
in new stack
-- Playing 'thank-you-for-calling' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, n01|s|1) in new stack
-- Goto (n01,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, n01) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-cooperation|m) in
new stack
-- Playing 'thank-you-cooperation' (language 'en')
== Spawn extension (n01, s, 2) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
I also tried Background(thank-you-cooperation|m||n01). The result is
- pressing 1 goto ext i
- pressing any other key does not stop sound playing, as expected.
the message on the manager interface when 1 pressing.
-- Starting simple switch on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Wait(Zap/1-1, 1) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-cooperation)
in new stack
-- Playing 'thank-you-cooperation' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Goto(Zap/1-1, n01|s|1) in new stack
-- Goto (n01,s,1)
-- Executing [EMAIL PROTECTED]:1] NoOp(Zap/1-1, n01) in new stack
-- Executing [EMAIL PROTECTED]:2] BackGround(Zap/1-1, thank-you-cooperation|m||n01)
in new stack
-- Playing 'thank-you-cooperation' (language '')
-- Sent into invalid extension 'E8' in context 'n01' on Zap/1-1
-- Executing [EMAIL PROTECTED]:1] Playback(Zap/1-1, digits/star) in new
stack
-- Playing 'digits/star' (language 'en')
== Auto fallthrough, channel 'Zap/1-1' status is 'UNKNOWN'
-- Hungup 'Zap/1-1'
NOTICE that * tries to go to ext 'E8' which is a French alphabet e with
grave accent.
DTMF detection problem? but if context option and m option of Background is
not specified, everything works well.
any help will be appreciated.
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[Asterisk-Users] can automon work with MixMonitor
Hi list automon now works as Monitor does. But MixMonitor is a better way for most cases, I guess. Any workaround to make automon do that? Any help will be appreciated. Franz Wu ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial cmd has delay for the last dialed number on FXO
Hi list
The * server one TDM04B card and my dialplan:
exten = 080.,1,Dial(Zap/g1/${EXTEN})
All four FXO ports has group=1 in zapata.conf
After dialing 0800012345 from a FXS extension, with one DTMF detector tapped
on the line, I found * dialed 0,8,0,0,0,1,2,3,4 in even interval, delay
longer, and then the last 5. Sometimes this behavior causes PSTN to repsond
with the number you dialed is non-existing.
Any help will be appreciated.
Franz
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Re: [Asterisk-Users] Installing the none commercial intel g729codecs into [EMAIL PROTECTED] 2.2?
Hi, Douglas Garstang wrote: Hang on there's a non commercial G729 codec that will work with Asterisk? Can someone point me to where I can find it? Check out http://www.voip-info.org/wiki-Asterisk+G.729+Licensing The binaries from http://kvin.lv/pub/Linux/Asterisk/ work for me (* 1.2.2 from svn, Debian), installing from source didn't. HTH, Franz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2?
Hi, MapsAir wrote: Has anyone successfully Installing the none commercial intel g729 codecs into [EMAIL PROTECTED] 2.2? Installed them today. Installing from source didn't work for me (Debian, Asterisk 1.2 from svn) but just adding the binaries (see the wiki on voip.org) did the job. Have you already tried the binaries? HTH, Franz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7912G SIP phone and Asterisk double RTP packets
Hi there, i did some tests with two Cisco 7912G phones (SIP stack) yesterday. With both ethereal and tcpdump listening on the Asterisk-Server's NIC, it came up that all RTP packets were doubled, with some small but almost constant delay (~460 us). The setup is 7912G -- ASTERISK -- 7912G The tcpdump output shows RTP traffic ASTERISK -- 7912G: 00 IP $ASTERISK.17944 $DEST.16384: udp/rtp 160 c0 4851 0 000279 IP $ASTERISK.17944 $DEST.16384: udp/rtp 160 c0 4851 0 006736 IP $ASTERISK.17944 $DEST.16384: udp/rtp 160 c0 4852 160 000460 IP $ASTERISK.17944 $DEST.16384: udp/rtp 160 c0 4852 160 015967 IP $ASTERISK.17944 $DEST.16384: udp/rtp 160 c0 4853 320 000460 IP $ASTERISK.17944 $DEST.16384: udp/rtp 160 c0 4853 320 019229 IP $ASTERISK.17944 $DEST.16384: udp/rtp 160 c0 4854 480 000458 IP $ASTERISK.17944 $DEST.16384: udp/rtp 160 c0 4854 480 019679 IP $ASTERISK.17944 $DEST.16384: udp/rtp 160 c0 4855 640 000459 IP $ASTERISK.17944 $DEST.16384: udp/rtp 160 c0 4855 640 ^^ time delta in micro-seconds ^^^ Sequence No ||| Timestamp Since sequence number and timestamp are equal for every two consecutive packets, and the payload is completely the same as i discovered using ethereal, it's obvious to me that this is one RTP stream, but two times sent. This doesn't occur using X-Lite. It's the same issue with SIP requests, the Cisco phones send two equal INVITES when you make a call (not so with X-Lite). My questions: Is this normal behaviour (i guess not)? What is the problem, the Cisco phones, Asterisk, tcpdump/ethereal...? Why are the calls not bridged between the two phones (RTP traffic just between the end-users) as they are when i use two X-Lite clients? I hope you have some answers to my problem :-) Thanks in advance, Franz ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] option r in Dial command seems not to work
hi list need your opinion. thanks in advance when calls come in from zap channel (te110p as E1/PRI) and go out to a SIP peer, no ringtone heard at zap channel. sip.conf: [from_sipproxy] progressinband=no type=peer context=from-proxy host=xxx.yyy.zzz.www port=5060 disallow=all allow=alaw extensions.conf [pri] exten = _X.,1,Dial(SIP/[EMAIL PROTECTED],,r) whether using option r in Dial command or not makes no diference. calls from sip to zap channel work well. i traced sip packets, found that sip device did send 180 Ringing. it looks like asterisk tells zap channel to say there's inband information but not to generate ringtone. on * console, pri debug span 1 shows Protocol Discriminator: Q.931 (8) len=47 Call Ref: len= 2 (reference 320/0x140) (Originator) Message type: SETUP (5) [a1]01*CLI Sending Complete (len= 1) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] [6c 0c 00 80 30 39 35 35 36 38 39 34 35 36] Calling Number (len=14) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation permitted, user number not screened (0) '0955689456' ] [70 0b 80 30 37 30 32 30 33 34 31 30 30] Called Number (len=13) [ Ext: 1 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) '0702034100' ] [7d 02 91 81] IE: High-layer Compatibility (len = 4) -- Making new call for cr 320 -- Processing Q.931 Call Setup -- Processing IE 161 (cs0, Sending Complete) -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) -- Processing IE 125 (cs0, High-layer Compatibility) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 320/0x140) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 8a] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 10 ] -- Executing Dial(Zap/10-1, SIP/[EMAIL PROTECTED]) in new stack -- Accepting call from '0955689456' to '0702034100' on channel 0/10, span 1 -- Called [EMAIL PROTECTED] -- SIP/220.228.43.182-9495 is ringing Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 320/0x140) (Terminator) Message type: ALERTING (1) [1e 02 81 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 320/0x140) (Originator) Message type: DISCONNECT (69) [08 02 80 90] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check Digium TE410P firmware version?
I connect to Asterisk via SSH all the times. Did not notice about console messages about module loading. Thanks - Original Message - From: BJ Weschke [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 10:24 Subject: Re: [Asterisk-Users] How to check Digium TE410P firmware version? On 12/26/05, Franz Wu [EMAIL PROTECTED] wrote: Hi list I have one TE410P and want to know how to. Sending back to Digium should be a good idea. When you load the wct4xxp module you'll see (1st Gen) or (2nd Gen) pop up which will indicate which version of the firmware the board is. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to check Digium TE410P firmware version?
Hi list I have one TE410P and want to know how to. Sending back to Digium should be a good idea. thanks in advance ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDMoE problem
Hi all my system 1: celeron 1.2GHz + intel 810e (asus TUW-LA) + 256MB SDRAM onboard vga (intel 810e chipset) RTL8100 NIC debian sarge 3.1r0a / kernel 2.6.8-2-686 asterisk / libpri / zaptel from CVS HEAD @ 2005-10-24 system 2: pentium II 533MHz + intel 810e (dfi PW35-E) + 256MB SDRAM onboard vga RTL8139D NIC TE110P debian sarge 3.1r0a / kernel 2.6.8-2-686 asterisk / libpri / zaptel from CVS HEAD @ 2005-10-24 CASE 1: configuring TDMoE and TE100P as the same time make kernel hang zaptel.conf of system 1 dynamic=eth,eth0/MAC#2,31,1 bchan=1-15,17-31 dchan=16 zapata.conf context = p1 swtichtype = qsig signalling = pri_cpe resetinterval = 3600 channel = 1-15 channel = 17-31 zaptel.conf of system 2 # for TDMoE dynamic=eth,eth0/MAC#1,31,0 bchan=1-15,17-31 dchan=16 # for TE110P span=1,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 zapata.conf context = e1 swtichtype = qsig signalling = pri_cpe resetinterval = 3600 channel = 1-15 channel = 17-31 context=p2 swtichtype = qsig signalling = pri_net resetinterval = 3600 channel = 32-46 channel = 48-62 with this configuration, modprobe ok. ztcfg - ok. but at system boot-up, the kernel dumps a lot of garbage (softirq.c badness). if the TE110P card removed from PCI slot of system 2 as well as corresponding config, things go well as long as * does not start. CASE II TDMoE span has a lot of frame reject and D-channel down and up all configuration same as CASE I except the TE110P removed. when more than 5 calls are set up between two systems, messages about frame reject, PRI got event: HDLC Bad FCS dumps on console and D-channel looks like going up and down and up and down. even with 5 or less calls, the sound quality is bad. On the voip-info.org wiki, somebody seems having same problem as me. Any opinion will be appreciated. Franz Wu ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDMoE and Badness in Kernel
2.6.13.4 which digium staff recommended. 2.6.14 both fail ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PSTN IAX Connections / Line Banks
I was wondering two things: I am running Asterisk Current Stable. Is there a way in Asterisk or using other equipment or through the TelCO that I can make a channel bank such that say I have 3 Phone number either with the use of FXO Cards or by using 3 connections via IAX to a service that could then provide PSTN connectivity where when Line 1 is called the call is then switched and held on line 2 or 3 such that Line 1 phone number is always free to receive calls. Secondly, does anyone know if there is a PSTN server provider that I can connect to via IAX protocol so I dont have to use a FXO device (I think this will provide better QoS) and also that has no service fee I just want to pay by the minute because it is going to be a FAX line only However it needs to have a number such that it can do both incoming and outgoing calls. Please get back to me if anyone knows anything thanks! Matthew H. Franz [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 4+ Port FXS Analog Device
Yeah you could use just a few analog ports and patch the phones to a patch panel and then have several phones on each FXS extension... However if you want them to all have their own ... I am not sure... Ill think about it, but right now the Cisco ATA or other FXS port at 1 FXS port / extension seems to be the only way I can think of... But again you could get like 5 ports and only have 5 extensions @ 4 phones / extension. Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Waldo Rubinstein Sent: Thursday, June 02, 2005 12:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 4+ Port FXS Analog Device I'm looking for an inexpensive way to connect 20 analog phones to asterisk. I could get a bunch of Linksys or Sipura boxes but was wondering if there is a more cost effective way? I came across the Mediatrix 1104 and even the Mediatrix 1124 but that comes out to be almost $100/port. I might as well buy inexpensive IP phone. Does anyone have any suggestions? Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Siemens optiPoint 420 phone and Asterisk
Dear Asterisk users, On Mon, 18 Apr 2005, Franz Knipp wrote: thanks for this information. I've contacted my customer adviser at Siemens, he'll try to organize me this version. I've got back the phones with a new SIP firmware. The following version informations are shown on the configuration page: Application: 4.1.17 SIP stack: 3.6.2.5 SIP signaling: 0.0.1 RTP: 0.0.11 Web content: 4.1.17 The phone administration page can be reached via https://phone ip/ The default administrator password is 123456, the user password 00. The web interface seems to work with Internet Explorer only, with Mozilla the administrator login box appears again and again. I've got two problems: * Calls between two phones of that kind have no voice transmission, signalling works. A connection from the Siemens phone to a Cisco SIP phone works without problems. * The labels of the function keys (which are small LCD displays) don't work yet. The web interface has the functionality to set them, but I receive an error when saving the changes. I can live with this limitations at the moment, but I'll be annoying to get new software versions as soon as they are available :-) Greetings from Vienna, Franz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NEED HELP PROGRAMING ASTERISK VoIP NETWORK
Please contact me Urgent... Atentamente, Franz Schuverer Arrue GLOBAL GROUP, INC. www.telefoniaglobal.net [EMAIL PROTECTED] Tel. (504) 221-4062 (Honduras Tel. (507) 322-2259 (Panamá) Tel. (866) 978-0976 (U.S.A.) (SKYPE) franz1969 (MSN MESSENGER) [EMAIL PROTECTED] (YAHOO MESSENGER) [EMAIL PROTECTED] CONFIDENCIALIDAD. El contenido de esta comunicación, así como el de toda la documentación anexa, es confidencial y va dirigido únicamente al destinatario del mismo. En el supuesto de que usted no fuera el destinatario, le solicitamos que nos lo indique y no comunique su contenido a terceros, procediendo a su destrucción. CONFIDENCIALITY. The content of this communication and any attached information is confidential and exclusively for the use of the addressee. If you are not the addressee, we ask you to notify to the sender and do not pass its content to another person, and please be sure you destroy it. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTERISK PROGRAMER
PLEASE CAN SOMBODY HELP ME PROGRAMIN AN VoIP ASTERISK SERVER Atentamente, Franz Schuverer Arrue GLOBAL GROUP, INC. www.telefoniaglobal.net [EMAIL PROTECTED] Tel. (504) 221-4062 (Honduras Tel. (507) 322-2259 (Panamá) Tel. (866) 978-0976 (U.S.A.) CONFIDENCIALIDAD. El contenido de esta comunicación, así como el de toda la documentación anexa, es confidencial y va dirigido únicamente al destinatario del mismo. En el supuesto de que usted no fuera el destinatario, le solicitamos que nos lo indique y no comunique su contenido a terceros, procediendo a su destrucción. CONFIDENCIALITY. The content of this communication and any attached information is confidential and exclusively for the use of the addressee. If you are not the addressee, we ask you to notify to the sender and do not pass its content to another person, and please be sure you destroy it. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: Sábado, 23 de Abril de 2005 11:00 a.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 9, Issue 209 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com 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: Cisco 7960 won't register as SIP device (List Receiver) 2. Re: if outgoing call fails with provider 1 then auto try provider 2 (Thomas Miller) 3. Re: if outgoing call fails with provider 1 then auto try provider 2 (Thomas Miller) 4. RE: Cisco 7960 won't register as SIP device (Robert Webb) 5. RE: Re: Hotel billing in IPSwitchBoard (Mathew McKernan) 6. Re: Hotel billing in IPSwitchBoard (Steve Rawlings) 7. RE: Cisco 7960 won't register as SIP device (Robert Webb) 8. RE: Cisco 7960 won't register as SIP device (List Receiver) 9. Re: Quadbri bristuff: can * respond only to 1 MSN and leave 1 number to other ISDN phones ? (Michiel van Baak) 10. Re: Hotel billing in IPSwitchBoard (tgj) 11. RE: Hotel billing in IPSwitchBoard (Chris Mason (Lists)) 12. [Fwd: FW: [Asterisk-Users] IAX help] (Michael DiMartino) 13. Re: Re: Hotel billing in IPSwitchBoard (tgj) 14. Re: OctoBRI and 2.6kernel (Michael Bielicki) 15. Re: [Fwd: FW: [Asterisk-Users] IAX help] (Peter Bowyer) 16. Re: Re: Hotel billing in IPSwitchBoard (David John Walsh) -- Message: 1 Date: Sat, 23 Apr 2005 08:23:32 -0700 From: List Receiver [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco 7960 won't register as SIP device To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii The DNS servers are valid. I configured the phone via .cnf files. The following are the sip.conf and sipMAC.cnf files. [tycisco] type=friend username=username secret=secret qualify=200 ; Qualify peer is no more than 200ms away nat=yes ;insecure=no host=dynamic; This device registers with us ;defaultip=24.18.147.95 canreinvite=no context=fullaccess dtmfmode=inband ;mailbox=101 disallow=all allow=ulaw allow=alaw allow=g729 .cnf: # SIP Configuration File (start) # Proxy Server proxy1_address: asterisk.mastermindpro.com proxy2_address: proxy3_address: proxy4_address: proxy5_address: proxy6_address: # Line 1 Settings line1_name: tycisco ; Line 1 Extension\User ID line1_displayname: 101 ; Line 1 Display Name line1_authname: username ; Line 1 Registration Authentication line1_password: secret ; Line 1 Registration Password # Line 2 Settings line2_name: ; Line 2 Extension\User ID line2_displayname:; Line 2 Display Name line2_authname: UNPROVISIONED ; Line 2 Registration Authentication line2_password: UNPROVISIONED ; Line 2 Registration Password # Line 3 Settings line3_name: ; Line 3 Extension\User ID line3_displayname:; Line 3 Display Name line3_authname: UNPROVISIONED ; Line 3 Registration Authentication line3_password: UNPROVISIONED ; Line 3 Registration Password # Line 4 Settings line4_name: ; Line 4 Extension\User ID line4_displayname:; Line 4 Display Name line4_authname: UNPROVISIONED ; Line 4 Registration Authentication line4_password: UNPROVISIONED ; Line 4 Registration Password # Line
[Asterisk-Users] Re: Siemens optiPoint 420 phone and Asterisk
Dear Richard, On Fri, 15 Apr 2005, [EMAIL PROTECTED] wrote: The latest firmware for optipoint420 advance SIP seems to be version 4.0.22A, released for HiPath8000. thanks for this information. I've contacted my customer adviser at Siemens, he'll try to organize me this version. What siemens PBX do you use? It's a HiPath 3300 (Rack version) with the extension containing 4 ISDN ports to connect to *. I don't know... maybe it will work... We only have several OptiPoint400 and they work fine. The risk of making the phone unuseable by installing a wrong firmware seems too high for me, so I won't try that. Thanks for the help! Bye, Franz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Siemens optiPoint 420 phone and Asterisk
Hi, today I've got two Siemens optiPoint 420 phones and I want to connect them to an existing Asterisk server. I didn't find any SIP firmware for that phone, according to announcements it will be released later this year (hopefully soon). chan_cornet, which would support the proprietary Siemens protocol, is not part of the CVS tree yet (thanks to Steffen Koepf for writing this). Maybe, someone of you can help me getting this phones working with Asterisk by pointing out a good starting point for my investigation and own development (if necessary) ;-) Last but not least, some kind of network diagram to clarify the situation: ISDN NT ---[Siemens PBX]--(S0)--[Asterisk]--(IP)--[optiPoint] Does anybody know, if it is worth trying out the optiPoint 400 SIP firmware on the 410/420 phones? Thanks for your assistance, greetings from Vienna, Franz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel compile error
Hi all, I am trying to compile Asterisk beginning with zaptel. Now I get 2 compile errors (see below). Can anyone give me a hint? Thanks Franz - sip:/usr/src/zaptel # make install cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o -lm -L. libtonezone.a cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -c ztspeed.c cc -o ztspeed ztspeed.o make -C /usr/src/linux-2.6 SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/linux-2.6.5-7.108' Makefile:438: .config: No such file or directory WARNING: Symbol version dump /usr/src/linux-2.6.5-7.108/Module.symvers is missing, modules will have CONFIG_MODVERSIONS disabled. CC [M] /usr/src/zaptel/zaptel.o /bin/sh: line 1: scripts/basic/fixdep: No such file or directory make[2]: *** [/usr/src/zaptel/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.5-7.108' make: *** [linux26] Error 2 sip:/usr/src/zaptel # ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as a Carrier SIP-PSTN Gateway
Hi all, I do not have any experience with Asterisk but I suppose that in principle it should be possible to use Asterisk as a pure Gateway between SIP and PSTN for carrier application. But maybe this is not the right equipment e.g. from reliability or administrative point of view. From PSTN traffic estimation there will be up to 100 E1 interfaces. Does anyone have any suggestions? Franz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Still problems at compiling
Hello experts, to avoid any unknown problems with my Linux installation I have now as a last resort method installed SuSE Linux 9.0 a new and have downloaded a fresh copy of Asterisk via CVS. Then I followed the steps of the Getting started with Asterisk and compiled successfully zaptel and libpri (as far as I can see). But when I compile asterisk I get an error. I have attached the sysout log below. Any hint and help highly appreciated. What is wrong. Franz the sysout log during make clean and make install of asterisk - linux:/usr/src/asterisk # make clean for x in res channels pbx apps codecs formats agi cdr astman stdtime; do make -C $x clean || exit 1 ; done make[1]: Entering directory `/usr/src/asterisk/res' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/res' make[1]: Entering directory `/usr/src/asterisk/channels' rm -f *.so *.o .depend rm -f busy.h ringtone.h gentone gentone-ulaw make[1]: Leaving directory `/usr/src/asterisk/channels' make[1]: Entering directory `/usr/src/asterisk/pbx' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/pbx' make[1]: Entering directory `/usr/src/asterisk/apps' rm -f *.so *.o look .depend make[1]: Leaving directory `/usr/src/asterisk/apps' make[1]: Entering directory `/usr/src/asterisk/codecs' rm -f *.so *.o .depend ! [ -d g723.1 ] || make -C g723.1 clean ! [ -d g723.1b ] || make -C g723.1b clean make -C gsm clean make[2]: Entering directory `/usr/src/asterisk/codecs/gsm' rm -f */*.o\ ./tst/lin2cod ./tst/lin2txt \ ./tst/cod2lin ./tst/cod2txt \ ./tst/gsm2cod \ ./tst/*.*.* find . \( -name core -o -name foo \) \ -print | xargs rm -f rm -f ./lib/libgsm.a ./add-test/add \ ./bin/toast ./bin/tcat ./bin/untoast\ ./gsm-1.0.tar.Z make[2]: Leaving directory `/usr/src/asterisk/codecs/gsm' make -C lpc10 clean make[2]: Entering directory `/usr/src/asterisk/codecs/lpc10' rm -f *.o ./liblpc10.a make[2]: Leaving directory `/usr/src/asterisk/codecs/lpc10' make -C ilbc clean make[2]: Entering directory `/usr/src/asterisk/codecs/ilbc' rm -f libilbc.a *.o make[2]: Leaving directory `/usr/src/asterisk/codecs/ilbc' make[1]: Leaving directory `/usr/src/asterisk/codecs' make[1]: Entering directory `/usr/src/asterisk/formats' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/formats' make[1]: Entering directory `/usr/src/asterisk/agi' rm -f *.so *.o look .depend make[1]: Leaving directory `/usr/src/asterisk/agi' make[1]: Entering directory `/usr/src/asterisk/cdr' rm -f *.so *.o .depend make[1]: Leaving directory `/usr/src/asterisk/cdr' make[1]: Entering directory `/usr/src/asterisk/astman' rm -f *.o astman .depend make[1]: Leaving directory `/usr/src/asterisk/astman' make[1]: Entering directory `/usr/src/asterisk/stdtime' rm -f libtime.a *.o test .depend make[1]: Leaving directory `/usr/src/asterisk/stdtime' rm -f *.o *.so asterisk .depend rm -f build.h rm -f ast_expr.c make -C db1-ast clean make[1]: Entering directory `/usr/src/asterisk/db1-ast' rm -f libdb1.a libdb.so.2 hash.o hash_bigkey.o hash_buf.o hash_func.o hash_log2.o hash_page.o ndbm.o bt_close.o bt_conv.o bt_debug.o bt_delete.o bt_get.o bt_open.o bt_overflow.o bt_page.o bt_put.o bt_search.o bt_seq.o bt_split.o bt_utils.o db.o mpool.o rec_close.o rec_delete.o rec_get.o rec_open.o rec_put.o rec_search.o rec_seq.o rec_utils.o hash.os hash_bigkey.os hash_buf.os hash_func.os hash_log2.os hash_page.os ndbm.os bt_close.os bt_conv.os bt_debug.os bt_delete.os bt_get.os bt_open.os bt_overflow.os bt_page.os bt_put.os bt_search.os bt_seq.os bt_split.os bt_utils.os db.os mpool.os rec_close.os rec_delete.os rec_get.os rec_open.os rec_put.os rec_search.os rec_seq.os rec_utils.os make[1]: Leaving directory `/usr/src/asterisk/db1-ast' make -C stdtime clean make[1]: Entering directory `/usr/src/asterisk/stdtime' rm -f libtime.a *.o test .depend make[1]: Leaving directory `/usr/src/asterisk/stdtime' linux:/usr/src/asterisk # make install ./mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -DASTERISK_VERSION=\CVS-01/20/04-10:14:14\ -DINSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\ -DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DBUSYDETECT_MARTIN -DNEW_PRI_HANGUP `ls *.c` cli.c:31:19: build.h: No such file or directory dlfcn.c:40:25: mach-o/dyld.h: No such file or directory dlfcn.c:41:26: mach-o/nlist.h: No such file or directory dlfcn.c:42:28: mach-o/getsect.h: No such file or directory for x in res channels pbx apps codecs formats agi cdr astman
RE: [Asterisk-Users] Still problems at compiling
From: Patrick Sent: Tuesday, January 20, 2004 2:29 PM Did you actually read the error message and try to understand solve the problem? No, being a Linux newbee and under a stress condition, I did not. But meanwhile I did and I installed several additional packages and now the compilation came to an end and brought an executable asterisk code. There were also various warnings during compilation which I generously ignored for this time Thanks for your patience with a stressed newbee. Patrick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] compiling problems
Hello, I still have problems with compiling Asterisk, and I am still on the first step at zaptel make clean; make install. I assume, that the troubles I have stem from a recent kernel-update I made. I upgraded from k_athlon_2.4.21_99_i586 to k_athlon_2.4.21_166_i586 via YaST Online Update.. Now I learned, that I have to provide also the kernel-sources for compiling zaptel. I have done that, but at the end of make install of zaptel I get the following errors on unresolved symbols: /sbin/depmod -a depmod: *** Unresolved symbols in /lib/modules/2.4.21-166-athlon/misc/wcusb.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-166-athlon/misc/zaptel.o depmod: *** Unresolved symbols in /lib/modules/2.4.21-166-athlon/misc/ztd-eth.o My experience on development tools is unfortunately very small. What do I have to do to move forward? Any help very appreciated. Franz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling problems with SuSE
From: Dustin Knuttgen on Sunday, January 18, 2004 11:47 PM We tried to use SuSE initially and had no luck compiling zaptel on either 8.2 or 9.0. We even had Digium take a look. After working on it for days we finally switched to Red Hat 9. Is there anyone who succeeded in compiling Asterisk with SuSE 8.2 or 9.0? Franz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem at compiling zaptel
Hi all!
Can anybody please give me some advice, what is wrong at my first try to
compile Asterisk. I have successfully downloaded the sources from CVS, but
now the next step at zaptel clean; make install fails.
Please have a look at the error-log below.
There must be a fundamental mis-configuration I suppose, but I am
unfortunately not an expert in this area.
Franz
-- error log -
lpc:/usr/src # cd zaptel
lpc:/usr/src/zaptel # make clean; make install rm -f torisatool makefw
tor2fw.h rm -f zttool rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f
zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo rm -f gendigits tones.h rm
-f libtonezone* rm -f tor2ee rm -f core
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB
-I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes
-fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I
/usr/src/linux/include -I/usr/src/linux/include/net -DSTANDALONE_ZAPATA -c
zaptel.c
In file included from /usr/include/linux/module.h:20,
from zaptel.c:44:
/usr/include/asm/module.h:54:2: #error unknown processor family In file
included from /usr/include/linux/mm.h:205,
from /usr/include/asm/pci.h:7,
from /usr/include/linux/pci.h:677,
from zaptel.c:46:
/usr/include/linux/page-flags.h:119: error: `CONFIG_X86_L1_CACHE_SHIFT'
undeclared here (not in a function)
/usr/include/linux/page-flags.h:119: error: requested alignment is not a
constant In file included from zaptel.c:48:
/usr/include/linux/version.h:2:2: #error
===
/usr/include/linux/version.h:3:2: #error You should not include
/usr/include/{linux,asm}/ header
/usr/include/linux/version.h:4:2: #error files directly for the compilation
of kernel modules.
/usr/include/linux/version.h:5:2: #error
/usr/include/linux/version.h:6:2: #error glibc now uses kernel header files
from a well-defined
/usr/include/linux/version.h:7:2: #error working kernel version (as
recommended by Linus Torvalds)
/usr/include/linux/version.h:8:2: #error These files are glibc internal and
may not match the
/usr/include/linux/version.h:9:2: #error currently running kernel. They
should only be
/usr/include/linux/version.h:10:2: #error included via other system header
files - user space
/usr/include/linux/version.h:11:2: #error programs should not directly
include linux/*.h or
/usr/include/linux/version.h:12:2: #error asm/*.h as well.
/usr/include/linux/version.h:13:2: #error
/usr/include/linux/version.h:14:2: #error To build kernel modules please do
the following:
/usr/include/linux/version.h:15:2: #error
/usr/include/linux/version.h:16:2: #error o Have the kernel sources
installed
/usr/include/linux/version.h:17:2: #error
/usr/include/linux/version.h:18:2: #error o Make sure that the symbolic
link
/usr/include/linux/version.h:19:2: #error/lib/modules/`uname -r`/build
exists and points to
/usr/include/linux/version.h:20:2: #errorthe matching kernel source
directory
/usr/include/linux/version.h:21:2: #error
/usr/include/linux/version.h:22:2: #error o Configure kernel sources:
/usr/include/linux/version.h:23:2: #error- cd /usr/src/linux
/usr/include/linux/version.h:24:2: #error- make mrproper
/usr/include/linux/version.h:25:2: #error- make cloneconfig
/usr/include/linux/version.h:26:2: #error- make dep
/usr/include/linux/version.h:27:2: #error
/usr/include/linux/version.h:28:2: #error o When compiling, make sure to
use the following
/usr/include/linux/version.h:29:2: #errorcompiler option to use the
correct include files:
/usr/include/linux/version.h:30:2: #error
/usr/include/linux/version.h:31:2: #error-I/lib/modules/`uname
-r`/build/include
/usr/include/linux/version.h:32:2: #error
/usr/include/linux/version.h:33:2: #errorinstead of
/usr/include/linux/version.h:34:2: #error
/usr/include/linux/version.h:35:2: #error-I/usr/include/linux
/usr/include/linux/version.h:36:2: #error
/usr/include/linux/version.h:37:2: #errorPlease adjust the Makefile
accordingly.
/usr/include/linux/version.h:38:2: #error
===
In file included from zaptel.h:36,
from zaptel.c:82:
/usr/include/linux/version.h:2:2: #error
===
/usr/include/linux/version.h:3:2: #error You should not include
/usr/include/{linux,asm}/ header
/usr/include/linux/version.h:4:2: #error files directly for the compilation
of kernel modules.
/usr/include/linux/version.h:5:2: #error
/usr/include/linux/version.h:6:2: #error glibc now uses kernel header files
from a well-defined
/usr/include/linux/version.h:7:2: #error working kernel version (as
recommended by Linus Torvalds)
/usr/include/linux/version.h:8:2
[Asterisk-Users] Problem at compiling zaptel (again)
Hi all!
Sorry, the error-log in my previous mail was disturbed.
Can anybody please give me some advice, what is wrong at my first try to
compile Asterisk. I have successfully downloaded the sources from CVS, but
now the next step at zaptel clean; make install fails.
Please have a look at the error-log below.
There must be a fundamental mis-configuration I suppose, but I am
unfortunately not an expert in this area.
Franz
-- error-log -
lpc:/usr/src # cd zaptel
lpc:/usr/src/zaptel # make clean; make install
rm -f torisatool makefw tor2fw.h
rm -f zttool
rm -f *.o ztcfg tzdriver sethdlc sethdlc-new
rm -f zonedata.lo tonezone.lo libtonezone.so.1.0 *.lo
rm -f gendigits tones.h
rm -f libtonezone*
rm -f tor2ee
rm -f core
cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -c -o
gendigits.o gendigits.c
cc -o gendigits gendigits.o -lm
./gendigits
gcc -I/usr/src/linux-2.4/include -O6 -DMODULE -D__KERNEL__ -DEXPORT_SYMTAB
-I/usr/src/linux/drivers/net -Wall -I. -Wstrict-prototypes
-fomit-frame-pointer -I/usr/src/linux/drivers/net/wan -I
/usr/src/linux/include -I/usr/src/linux/include/net -DSTANDALONE_ZAPATA -c
zaptel.c
In file included from /usr/include/linux/module.h:20,
from zaptel.c:44:
/usr/include/asm/module.h:54:2: #error unknown processor family
In file included from /usr/include/linux/mm.h:205,
from /usr/include/asm/pci.h:7,
from /usr/include/linux/pci.h:677,
from zaptel.c:46:
/usr/include/linux/page-flags.h:119: error: `CONFIG_X86_L1_CACHE_SHIFT'
undeclared here (not in a function)
/usr/include/linux/page-flags.h:119: error: requested alignment is not a
constant
In file included from zaptel.c:48:
/usr/include/linux/version.h:2:2: #error
===
/usr/include/linux/version.h:3:2: #error You should not include
/usr/include/{linux,asm}/ header
/usr/include/linux/version.h:4:2: #error files directly for the compilation
of kernel modules.
/usr/include/linux/version.h:5:2: #error
/usr/include/linux/version.h:6:2: #error glibc now uses kernel header files
from a well-defined
/usr/include/linux/version.h:7:2: #error working kernel version (as
recommended by Linus Torvalds)
/usr/include/linux/version.h:8:2: #error These files are glibc internal and
may not match the
/usr/include/linux/version.h:9:2: #error currently running kernel. They
should only be
/usr/include/linux/version.h:10:2: #error included via other system header
files - user space
/usr/include/linux/version.h:11:2: #error programs should not directly
include linux/*.h or
/usr/include/linux/version.h:12:2: #error asm/*.h as well.
/usr/include/linux/version.h:13:2: #error
/usr/include/linux/version.h:14:2: #error To build kernel modules please do
the following:
/usr/include/linux/version.h:15:2: #error
/usr/include/linux/version.h:16:2: #error o Have the kernel sources
installed
/usr/include/linux/version.h:17:2: #error
/usr/include/linux/version.h:18:2: #error o Make sure that the symbolic
link
/usr/include/linux/version.h:19:2: #error/lib/modules/`uname -r`/build
exists and points to
/usr/include/linux/version.h:20:2: #errorthe matching kernel source
directory
/usr/include/linux/version.h:21:2: #error
/usr/include/linux/version.h:22:2: #error o Configure kernel sources:
/usr/include/linux/version.h:23:2: #error- cd /usr/src/linux
/usr/include/linux/version.h:24:2: #error- make mrproper
/usr/include/linux/version.h:25:2: #error- make cloneconfig
/usr/include/linux/version.h:26:2: #error- make dep
/usr/include/linux/version.h:27:2: #error
/usr/include/linux/version.h:28:2: #error o When compiling, make sure to
use the following
/usr/include/linux/version.h:29:2: #errorcompiler option to use the
correct include files:
/usr/include/linux/version.h:30:2: #error
/usr/include/linux/version.h:31:2: #error-I/lib/modules/`uname
-r`/build/include
/usr/include/linux/version.h:32:2: #error
/usr/include/linux/version.h:33:2: #errorinstead of
/usr/include/linux/version.h:34:2: #error
/usr/include/linux/version.h:35:2: #error-I/usr/include/linux
/usr/include/linux/version.h:36:2: #error
/usr/include/linux/version.h:37:2: #errorPlease adjust the Makefile
accordingly.
/usr/include/linux/version.h:38:2: #error
===
In file included from zaptel.h:36,
from zaptel.c:82:
/usr/include/linux/version.h:2:2: #error
===
/usr/include/linux/version.h:3:2: #error You should not include
/usr/include/{linux,asm}/ header
/usr/include/linux/version.h:4:2: #error files directly for the compilation
of kernel modules.
/usr/include/linux/version.h:5:2: #error
/usr/include/linux/version.h:6:2: #error glibc now uses kernel header files
from a well-defined
/usr/include/linux/version.h:7:2: #error working kernel version
RE: [Asterisk-Users] linux journal article on asterisk
From: [EMAIL PROTECTED] Sent: Wednesday, January 14, 2004 For anybody who didn't know there is an article on asterisk in February's Linux Journal. Can you please provide a link to this article? Franz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie ... some questions
Hi guruz, I haverequirements from a company, which isgoing to deploy call center nationwide using asterisk and the new 4 port cards. However before going to purchase thehardware they want if following is possible in the asterisk software 1) they want to whisper withone side of the call i.e. the manager while monitoring thecalls (from customers to support staff) from his extension (either SIP or Zap Channel),canguide support personif he is in trouble talking with the client 2) while monitring the call, incharge can take the call and start talking with the customer directly and the support officer gets a hangup tone Plz suggest me if the above is possible and how the above can be achieved. TIA Franzi Post your free ad now! Yahoo! Canada Personals
