Re: [asterisk-users] 603 Error
2 apr 2007 kl. 10.16 skrev Dovid B: Hi Guys, I started getting this error only from one of our ITSP's once we upgraded from 1.2.16 to 1.2.17. Can anyone shed light ? --- (12 headers 0 lines) --- Transmitting (NAT) to 209.212.93.53:5060: SIP/2.0 603 Declined (no dialog) Via: SIP/2.0/UDP XXX.XXX.XX.XX;branch=z9hG4bKf928.2b3b0de5.0;received=XXX.XXX.XX.XXX Via: SIP/2.0/UDP XXX.XXX.XX.XXX: 5060;rport=5060;received=74.96.44.239;branch=z9hG4bK-24c7d466 From: sip:XXX.XXX.XX.XX;tag=a21dc3d8dd92817bo0 To: sip:XXX.XXX.XX.XX;tag=as7b187bff Call-ID: [EMAIL PROTECTED] CSeq: 112226 NOTIFY User-Agent: Blah Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 Your server is sending a NOTIFY that the ITSP's server doesn't like. Propably a mailbox notification. Not a critical error, just a configuration issue. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk SIP master class, Stockholm may 2007 - register now! ___ --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] SIP - Automatic Redial on No Answer
2 apr 2007 kl. 10.46 skrev Olivier: Hi, What is the best way to implement Automatic Redial on No Answer ? Looking at http://www.ietf.org/internet-drafts/draft-ietf-sipping- service-examples-12.txt I can see how Automatic Redial on Busy could (should) be done. How would you do it on No Answer ? Is there any event you should SUBSCRIBE to so that you're notified that you're callee is available ? What if you ask to be notified of next call ending ? This is particularly useful when phones accept several calls : on some of them, I couldn't find a way to force them to reply 486 BUSY after a wait, when the callee is on call and couldn't explicitly reject or accept the incoming call. The scenario is : A calls B which is already on call, B is notified another call is here but B don't either reject or answer the incoming call A has no way to know if the call is not answered because B is not there or too busy to reply Ideally, A would then ask for Automatic Redial on No Answer, with a NOTIFY-SUBSCRIBE Later, when B is finishing a call (the one he was busy with or a brand new one after returning to his desk), A is notified and Automatic Redial can occur. As always, there are many ways to handle this. If your phone supports SUBSCRIBE, Asterisk does it too. In that case, it's up to the phone. Otherwise, you could come up with a multiprotocol dialplan-based solution where you - answer A's call and play a prompt for busy or no reply - like in Voicemail - ask the user to press 1 to redial (if B is busy) - have an app monitor B's connection over manager - when B is done, place call to A, then connect to B - lock B so no one else calls in between The big issue here is the external app. Maybe someone can figure out a way to do it without it... /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk SIP master class, Stockholm may 2007 - register now! ___ --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] LDAP authentication in Asterisk
2 apr 2007 kl. 13.50 skrev sravana: Anybody done LDAP authentication in Asterisk? can you explain how? Thanks in advance There's some code available in the issue tracker. Please check in bugs.digium.com for res_auth /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk SIP master class, Stockholm may 2007 - register now! ___ --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] SDP bug
2 apr 2007 kl. 19.32 skrev Raj Jain: I found a subtle difference between the two traces you sent (the call that works and the call that gets dropped). This may or may not be what's causing the problem. The call that gets dropped had a retransmission of INVITE from UAC to UAS (and therefore retransmission of 200 OK from UAS to UAC). There is nothing wrong with the re-transmission as such, but I noticed a potential bug in Asterisk in the way it responds to an INVITE retransmission. Asterisk is bumping up the session version number in the retransmitted 200 OK's SDP. This is as if Asterisk is treating the INVITE retransmission as a RE-INVITE. Asterisk sends 200 OK: o=root 16300 16300 IN IP4 203.89.nnn.nnn Asterisk sends 200 OK (retransmission): o=root 16300 16301 IN IP4 203.89.nnn.nnn Ideally, this bug should have nothing to do with why Asterisk is ignoring the ACK (which is why it keeps reatrasmitting the 200 OK and eventually drops the call). However, if you can confirm that all dropped calls have INVITE retransmission then that might give us a clue? Raj, That's an interesting observation. Do you think this will cause any issues? Even though it's not beautiful, I fail to see why a UA would check that. /O ___ --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] Re: On Topic: Cheapest Asterisk USB Key?
On Mon, 2007-04-02 at 22:12 -0400, Matthew Rubenstein wrote: What it means is that Flash memory cells wear out after a large number of read/write cycles, but not nearly as large as hard drives: http://en.wikipedia.org/wiki/Flash_rom#Limitations . So using Flash in place of RAM, even when high speed isn't important, can wear out the Flash - it will probably wear out even before HDs, which live less long than does RAM. Until the Flash wears out, it is extremely reliable, and techniques for ensuring it doesn't destroy data as it wears out are built into the Flash HW (though it will eventually wear out take data with it). But I'm not talking about using the Flash as RAM, just using it for a low-load persistent store like a HD, where a HD would be overkill in every way. I thought flash waers out on writing, not reading... So, keep /tmp,/var/log and its friends on ram-disk, or pass the logging you don't want to loose via a remote syslog, a remote mysql-server (or via nfs-mount). (For running servers I keep /usr and /opt mounted as read-only, to avoid accidental writing) -- pgp-id: 926EBB12 pgp-fingerprint: BE97 1CBF FAC4 236C 4A73 F76E EDFC D032 926E BB12 Registered linux user: 75761 (http://counter.li.org) ___ --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] Re: On Topic: Cheapest Asterisk USB Key?
On Mon, 2 Apr 2007, Matthew Rubenstein wrote: But I'm not talking about using the Flash as RAM, just using it for a low-load persistent store like a HD, where a HD would be overkill in every way. I boot my systems off a flash IDE drive. There's a partition with just enough of a root filesystem to make lilo work, a /boot directory with a kernel image and memtest and an initrd.gz which is a compressed ext2 filesystem... It is uncompressed into RAM and then the system runs entirely from RAM. There is a small 2nd partition on the device which I keep a tar-file of configuration settings. This us untarred once the system boots and fills in things like /etc/asterisk, /var/www/docs and a few other config files, including /var/spool/asterisk/astdb. I have a 2nd flash IDE drive for voicemail. This is mouinted as a live filesystem and I use GSM only to store voicemail (so a 64MB device is going to give me many hours of VM storage). Seems to work for me and keeps writes back to the important flash device (the boot one) to a bare minimum... I force fsck on the voicemail device at boot time, if there are any errors and I'm working on a 'sanitiser' too, which will remove any broken files - so it'll make sure there is a .WAV file for every .txt file and so on. I figure losing the occasional voicemail might be acceptable after someone pulls the plug on it. (and my experiences of this have been good in that I've nver yet lost a file) Gordon ___ --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] Replicating SIP Registrations Across Asterisk Servers
2 apr 2007 kl. 21.43 skrev John C. Wolosuk Jr.: Does any one know if there's an mechanism (internal to asterisk or otherwise) to replicate dynamic SIP device registrations across a pool of asterisk servers? I'm in the process of creating a asterisk cluster using a SIP hardware load balancer and so far this is one of the challenges I'm facing. One thought I'm currently investigating is to use openSER to intercept and replicate the incoming SIP REGISTER packets to all servers... The other thought in the back of my mind is to completely removing the task of handling registrations from asterisk and give it to SER directly or other registrar server. Using the realtime subsystem, you can share registration data between Asterisk servers. In combination with Dundi and the regexten= system, it's even more dynamic. /Olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Asterisk SIP master class, Stockholm may 2007 - register now! ___ --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] SIP 484 (Early Dial) and International Dialing
2 apr 2007 kl. 22.21 skrev James FitzGibbon:
I'm building a dialplan for use with a bunch of GXP2000 desk sets.
During testing, we had some user issues surrounding the lack of an
on-phone dialplan. Users would hit 9 and sit there waiting for a
redial tone, and the GXP would time out, sending just '9' to *,
which couldn't do much other than spit back a 404 or play pbx-invalid.
I turned on the early dial option on the GXP, which causes each
digit to be sent as it is pressed, and the user response was much
more favourable. Now I come to set up my international dialplans
and I'm running into a problem.
The textbook dial pattern for international calls:
_9011.
Isn't working because * matches the first digit after 011 and sends
an incomplete dialstring (dialing something like Zap/R1/0119 for
example).
I've tried using patterns with multiple . wildcards, and switching
from . to X, putting patterns like
_9011XXX
_9011XX
_9011X
In the hopes that * would see that 90119 could potentially match
a longer extension and not match immediately. No luck though -
dialing still starts immediately when one digit past 011 is received.
Any thoughts on how to get around this? Right now the best I have
(and that's not saying much) is to have something like:
[initialcontext]
exten = _9011,1,DISA(no-password|somecontext)
[somecontext]
exten = _X.,1,Dial(Zap/R1/011${EXTEN})
But that's ugly, not to mention confusing to the users because the
amplitude of the dialtone generated by the GXP is lower than the
dialtone generated by *, so they notice the bump when they've
dialed 9011.
When SIP sends an INVITE, it's a complete INVITE. The dialstring in
the invite is done and can't be added to, unless you have
enabled overlap dialling in SIP. When the phone sends a number, we
match and set up the call or fail.
Overlap dialling in SIP works by testing the dialstring. If it's not
an exact match, Asterisk will send a SIP response saying that
it needs more digits to determine the destination. In 1.4, this is
disabled by default and needs to be enabled.
You are assuming that SIP works like zaptel in the dialplan, but it
does not. You propably need to re-configure your phones.
/O
---
* Olle E. Johansson - [EMAIL PROTECTED]
* Asterisk Training http://edvina.net/training/
* Asterisk SIP master class, Stockholm may 2007 - register now!
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Re: [asterisk-users] chan-capi-HEAD and Asterisk 1.4.2
On Mon, 2 Apr 2007, Peer Oliver Schmidt wrote: Hello Armin, [EMAIL PROTECTED]:~# grep capi /var/log/asterisk/messages [Mar 31 16:17:03] ERROR[3850] chan_capi.c: Unable to listen on contr1 (error=0x100f) Is this helpful, or do you need more information? Yes, at this state it might be possible that less CPU power causes problems. The 'listen' command expects an answer and maybe it is coming too late. Can you please try the patch below? Index: chan_capi.c === --- chan_capi.c (revision 436) +++ chan_capi.c (working copy) @@ -631,7 +631,7 @@ error = LISTEN_CONF_INFO(CMSG); break; } - usleep(2); + usleep(10); tried the patch, but it did not work. It waits quite a long time before the chan-capi error message comes up, according to the time stamp it is about 12 seconds. It is kind of strange, that the whole startup process for asterisk usually takes only about 4-5 seconds. That's too long, normaly the confirmation message arrives within a few msecs. So it seems that the driver isn't responding. Do you need additional information? Which card/driver do you use? A debug log (capi trace) from the driver or kernelcapi helps to see what messages are wrong/missing. Armin ___ --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] freepbx - DB Error messages...
On Sat, March 24, 2007 19:10, Bruce Reeves wrote: You might get a faster response on freepbx/amp mailing list. On 3/24/07, Francesco Peeters (Asterisk) [EMAIL PROTECTED] wrote: SNIP Just an update: Still have NOT been approved for either the mailing list *or* the forum! I am pretty disappointed in the moderators! If you take up the responsibility to moderate a list or forum you have to make sure you respond promptly, especially if the list or forum (or both) require moderator approval before a user-account is activated! (And no, my original answer has not been answered yet either!) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=0 sync_slave=0 timer_card=0 VIA EPIA V8000 - 256 MB - * 1.2.4 - mISDN, but still no freePBX 2 Sweex HFC-PCI cards ___ --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] Re: asterisk-users Digest, Vol 33, Issue 9
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] SIP - Automatic Redial on No Answer
Hello, I've wrote a dialplan script which uses the H extension to do something similar to what you want. In general it uses the internal ASTDB for this: - When there is no answer (or busy) the caller hangs up, initiate a new call with some special code (*41 is used here by the public carrier, so I am using it also). Asterisk registers the data in its DB. - When the user disconnects the H extension is called. It then looks in ASTDB to see whether there is a user camoing on this extension. If so, a call file is created and Asterisk initiates the call. If this is what you need please tell me and I'll post the code on Thursday. I've already posted it in the past so you might search the archives in the meantime. __Yehavi: ___ --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] understanding what h extension does
I am trying to make a dialplan that when I dial 90 I can go round a
whole set of extensions and leave them a short message, hangup and go
on the next one.
I use the M facility of dial, with something like this
[messages]
exten = 90,n(calcnextchan),Set(DIALCHAN=...)
exten = 90,n,Dial(${DIALCHAN},30,M(domessage))
exten = 90,n,Goto(calcnextchan)
[macro-domessage]
exten = s,1,Playback(message)
exten = s,n,Set(MACRO_RESULT=CONTINUE)
[There is actually more logic to check for busy dial channels and retry
them later]
This seems to work fine until one of the callees hangs up before the
message is played. at which point my call is terminated.
I was wondering if I should user the h extension here to pickup the
hungup call from the callee and continue. However I am worried that I
might end up looping if I hang up my end of the call - since I want it
to stop if I do that.
I can't find a definitive explanation of what causes the h extension to
be called. Can someone explain what what happen if I added something
like
exten = h,1,Goto(90,calcnextchan)
to the [messages] context
--
Alan Chandler
http://www.chandlerfamily.org.uk
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Re: [asterisk-users] adding chan_celliax
Ciao Patricio, have you compiled Asterisk from sources? At the moment you can only add chan_celliax support if you compiled from source. If this is the case, I can give you full instruction. Giovanni On 4/3/07, Patricio Valarezo Lozano [EMAIL PROTECTED] wrote: Hi, I've installed asterisk 1.2 on debian/unstable and i would like to add the chan_celliax channel to my existing configuration in debian, is there a way to do that?? -- patoVala Linux User#280504 Hablando en http://www.elprimoalcahuete.com markm c++: the power, elegance and simplicity of a hand grenade ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan-capi-HEAD and Asterisk 1.4.2
Good Morning Armin, tried the patch, but it did not work. It waits quite a long time before the chan-capi error message comes up, according to the time stamp it is about 12 seconds. It is kind of strange, that the whole startup process for asterisk usually takes only about 4-5 seconds. That's too long, normaly the confirmation message arrives within a few msecs. So it seems that the driver isn't responding. Do you need additional information? Which card/driver do you use? [EMAIL PROTECTED]:~# lspci -s 0:0e -v 00:0e.0 Network controller: AVM Audiovisuelles MKTG Computer System GmbH A1 ISDN [Fritz] (rev 02) Subsystem: AVM Audiovisuelles MKTG Computer System GmbH FRITZ!Card ISDN Controller Flags: medium devsel, IRQ 10 Memory at ff001400 (32-bit, non-prefetchable) [size=32] I/O ports at dcc0 [size=32] [EMAIL PROTECTED]:~# capiinit status 1 fcpci running fcpci-dcc0-10A1 3.11-07 0xdcc0 10 Driver from ubuntu edgy A debug log (capi trace) from the driver or kernelcapi helps to see what messages are wrong/missing. What is the best way to produce this? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] SDP bug
Olle, It depends on how strictly the UA adheres to the offer/answer model. The issue would be that a RE-INVITE from Asterisk will have the version number incremented by more than one, which will break the following rule. Quoting from RFC 3264 Section 8: When issuing an offer that modifies the session, the o= line of the new SDP MUST be identical to that in the previous SDP, except that the version in the origin field MUST increment by one from the previous SDP. That said, I agree that most UAs do not check this. What's a bit more alarming fundamentally is that Asterisk is creating a new answer SDP to respond to an INVITE retransmission. An RFC 3261 compliant implementation MUST send an exact copy of the previous SIP response. Anyway, I realize that Asterisk is not inherently RFC 3261 compliant. Raj Asterisk sends 200 OK: o=root 16300 16300 IN IP4 203.89.nnn.nnn Asterisk sends 200 OK (retransmission): o=root 16300 16301 IN IP4 203.89.nnn.nnn Raj, That's an interesting observation. Do you think this will cause any issues? Even though it's not beautiful, I fail to see why a UA would check that. /O ___ --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] tdm400 problem
Hi all I have a problem with an tdm400 with 2 modules 1 fxo 1 fxs it just doesnt load the fxs module i dunno why... zaptel.conf loadzone=es defaultzone=es # qozap span definitions # most of the values should be bogus because we are not really zaptel span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 fxsks=13 fxoks=14 zapata.conf [channels] language=es echocancel=yes context=from-pstn echocancel=yes echocancelwhenbridged=yes echotraining=500 rxgain=-4.0 txgain=-6.0 usecallerid=yes hidecallerid=no threewaycalling=yes ;; RDSI BRI switchtype = euroisdn signalling = bri_cpe group=0 channel = 1-2,4-5,7-8 #channel = 1-2,7-8 ;; FXO signalling=fxs_ks group=1 rxgain=1.0 txgain=-6.0 busydetect=yes channel = 13 ;; FXS signalling=fxo_ks group=2 context=from-internal channel = 14 -- dmesg Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXO (FCC mode) !!! LOOP_CLOSE_TRES iREG 1C = 1 should be 1000 !!! RING_TRIP_TRES iREG 1D = 8000 should be 3600 !!! COMMON_MIN_TRES iREG 1E = 0 should be 1000 !!! COMMON_MAX_TRES iREG 1F = 0 should be 200 !!! PWR_ALARM_Q1Q2 iREG 20 = 1480 should be 7C0 !!! PWR_ALARM_Q3Q4 iREG 21 = 37C0 should be 2600 !!! PWR_ALARM_Q5Q6 iREG 22 = 3D70 should be 1B80 !!! LOOP_CLOSURE_FILTER iREG 23 = 3970 should be 8000 !!! RING_TRIP_FILTER iREG 24 = 78E0 should be 320 !!! TERM_LP_POLE_Q1Q2 iREG 25 = 8B60 should be 8C !!! TERM_LP_POLE_Q3Q4 iREG 26 = 6A40 should be 100 !!! TERM_LP_POLE_Q5Q6 iREG 27 = 8070 should be 10 !!! CM_BIAS_RINGING iREG 28 = should be C00 !!! DCDC_MIN_V iREG 29 = should be C00 !!! DCDC_XTRA iREG 2A = should be 1000 !!! LOOP_CLOSE_TRES_LOW iREG 2B = should be 1000 ! Init Indirect Registers UNSUCCESSFULLY. Indirect Registers failed verification. Proslic Failed on Second Attempt to Calibrate Manually. (Try -DNO_CALIBRATION in Makefile) Module 1: FAILED FXS (FCC) Module 2: Not installed Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Registered tone zone 6 (Spain) -- [Jorge J. Boscán Etura] quando omni flunkus moritatus Linux 2.6.17 X86_64 running fc6, lu #137000 +34636029900 ___ --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] SDP bug
The call that gets dropped had a retransmission of INVITE from UAC to UAS (and therefore retransmission of 200 OK from UAS to UAC). There is nothing wrong with the re-transmission as such, but I noticed a potential bug in Asterisk in the way it responds to an INVITE retransmission. Asterisk is bumping up the session version number in the retransmitted 200 OK's SDP. This is as if Asterisk is treating the INVITE retransmission as a RE-INVITE. Asterisk sends 200 OK: o=root 16300 16300 IN IP4 203.89.nnn.nnn Asterisk sends 200 OK (retransmission): o=root 16300 16301 IN IP4 203.89.nnn.nnn Ideally, this bug should have nothing to do with why Asterisk is ignoring the ACK (which is why it keeps reatrasmitting the 200 OK and eventually drops the call). However, if you can confirm that all dropped calls have INVITE retransmission then that might give us a clue? Raj, That's an interesting observation. Do you think this will cause any issues? Even though it's not beautiful, I fail to see why a UA would check that. I have run a number of tests and in all cases the calls that fail have a retransmitted INVITE whereas the successfull calls have only one INVITE. Regards Cameron ___ Now you can scan emails quickly with a reading pane. Get the new Yahoo! Mail. http://uk.docs.yahoo.com/nowyoucan.html ___ --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] misdn and debian
Monday, April 2, 2007, 7:30:57 PM, Giedrius wrote: Has anybody debian and misdn working fine? Maybe you can advices , what kernel and misdn versions to use... I use kernel 2.6.20.1 and misdn 1.1.0 with fritz card, and working fine. The kernel, asterisk (1.2.15) and misdn also compiled from source. -- Best regards, Gergomailto:[EMAIL PROTECTED] ___ --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] DTMF problem with 1.4.1
You can use the following to display what you receive from user (dtmf): exten= 1,1,Read(test) exten= 1,2,NoOp(DTMF Received: $test) exten= 1,3,Hangup On 4/3/07, Nitin Gupta [EMAIL PROTECTED] wrote: I upgraded to 1.4.1 and my DTMF has stopped working, I tried rfc2833compensate=yes and relaxdtmf=yes etc but none working. Everything seems to work fine with 1.2.10 Is there any way I dump the dtmf data packets received by asterisk on console? Any idea or pointers to debug the issue will be much appreciated. thanks, Nitin ___ --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 -- Regards Rizwan Hisham Software Engineer ___ --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] misdn and debian
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Have you got zaptel installed on your box? And loading the zaphfc module during bootup? I discovered the problem you reported when I switched to mISDN and having installed/loading zaphfc during bootup. Than asterisk doesn't start and the system hangs. I've several machines running on Debian with asterisk and mISDN without problems. Chris... Giedrius Augys schrieb: Hi, I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable debian 3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it stops near Apache2 starting I started my system with recovery kernel, and tun off misd, then my system works fine. I think it's problem with memory. Has anybody debian and misdn working fine? Maybe you can advices , what kernel and misdn versions to use... 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 - -- Dipl.-Ing. Kurt Krenn - IT-Beratung Franz-Josef-Strasse 33/4/43, 5020 Salzburg Tel: +43 662 879512 Fax: +43 662 875960 IP-Tel: +43 780 kkrenn (557366) Email: [EMAIL PROTECTED] sip: [EMAIL PROTECTED] -BEGIN PGP SIGNATURE- Version: GnuPG v2.0.3 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGEhs3R0exH8dhr/YRAn5oAJ4xOLKjyZ0p3iZcn6SN/XgjwGSfUACgnMv6 btwewQ5RYFMyLv01e+fwfys= =k7JJ -END PGP SIGNATURE- ___ --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] Quad BRI cards
Hi, I have a couple of questions about Quad-BRI solutions for Asterisk, and was hoping that I might get some feedback based on other people's experience. We currently use the Junghanns card, which is a pure Zaptel solution, which is fantastic, but they have no hardware EC solution, and their drivers are becoming increasingly un-stable with time (I back-port to a modified qozap driver from 0.2.0-RC8n which is the last one I can run without bad behaviour) I am aware of the Beronet BN4S0, which appears to be exactly the same card as the Junghanns card, but with mISDN drivers, still no h/w EC solution, and as a result, less effective (from what I read here) software echo cancellation. I thought I had struck gold when I saw the Digium B410P, which had a driver that builds as part of Zaptel, but then when I read on the list people describe it as an mISDN based card... Which is it? I prefer a ZAP based driver because I use that for the Sangoma and Single-BRI solutions that we build. I assume that if I switch to mISDN, I will need to install all of the Linux ISDN support, change my dialplan to use CAPI/ as a technology, use new and unfamiliar config files, and all sorts of other horribleness, probably losing one or two ZAP/ specific features such as the ZapEC() command in the process? Perhaps there is an alternative solution that I have missed entirely out there? The Single BRI (HFC) card has the vzaphfc alternative driver available, has anyone done the same for the Quad (HFC4S) card? Thanks for any pointers that can be provided. Kind regards, Steve ___ --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 Macros
Hello,
I've seen this already asked and answered but it is still a no go for
me.
I'm trying to do some preprocessing in the middle of a call, before
bridging.
I've seen two choices: M() and G() parameters of the Dial() command.
G() was discarded because I don't know if it is possible to bridge
channels after processing.
With M() I've done something like that:
macro screen ( screen_file, destination, caller_email ) {
Set(screen_file=${ARG1});
Set(destination=${ARG2});
Set(caller_email=${ARG3});
begin:
// compute play prompt for background()
Set(BACKGROUND_PROMPT=voip-call-pending${screen_file});
Wait(0.5);
Background(${BACKGROUND_PROMPT});
catch 2 {
Noop(GOTCHA!!);
};
catch t {
goto s|begin;
};
};
The dial command looks like this:
Dial(IAX2/shortcut1:[EMAIL PROTECTED]/[EMAIL PROTECTED]|120|M
(screen^${SCREEN_FILE}^${EXTEN}^${EMAIL_ADDRESS}));
What I do want is to ask the called person to press a key and make a
choice.
everything goes well until a key is pressed, macro exits with status
48 + ASCII code of the key and the call is bridged.
I've read that for both G() and M() the pbx services are not
available. This means that I cannot read a DTMF option in the
Background() command during the called person IVR?
Shouldn't catch 2 { } block catch the press of the '2' key and print
GHOTCHA!! ?
Thanks,
Alex
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[asterisk-users] Re: Asterisk and Fax
uxbod == [EMAIL PROTECTED] writes: uxbod Hi, I have a requirement for sending and receiving faxes and uxbod was wondering the best way to achieve it with Asterisk as I uxbod only have one phone line. uxbod I currently have a TDM11B in my server (1 x FXO, 1 x FXS), so I uxbod was thinking that I would need to get a additional FXS module, uxbod connect that to a Eicon Fax card, and then when receiving a uxbod call detect the fax tone and bridge the call to the FXS uxbod channel. I have had perfect luck so far with iaxmodem and Hylafax. I don't know how well it works with analog lines though. /Benny ___ --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] Zaptel 1.4.1 Install Modules CentOS
Hi All, I have a CentOS server that I am trying to configure Asterisk on 1.4 on. Everything seems to go ok, with regards to compiling Zaptel, Libpri, Asterisk (will be using kernel 2.6 timer and ztdummy) Unfortunately I can't insmod / modprobe ztdummy. [root @xyz src]# modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy [EMAIL PROTECTED] src]# insmod ztdummy insmod: can't read 'ztdummy': No such file or directory This is really causing me to scratch my head, the timer module is loaded ok, I simply don't know what is going wrong with the modules? I'm a bit out of my depth with CentOS, as this isn't my server (I'm a Slackware guy) Any pointers seriously appreciated. Thanks Chris -- Chris Blunt ___ --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] Zaptel 1.4.1 Install Modules CentOS
Chris Blunt wrote: I have a CentOS server that I am trying to configure Asterisk on 1.4 on. Everything seems to go ok, with regards to compiling Zaptel, Libpri, Asterisk (will be using kernel 2.6 timer and ztdummy) Unfortunately I can't insmod / modprobe ztdummy. Did you yum install kernel-devel-`uname -r` ? # Install dev tools: yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf libtool make automake automake14 automake15 automake16 automake17 bison byacc flex libtermcap libtermcap-devel newt newt-devel ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel # Zaptel: cd /usr/src/ \ wget -c http://ftp.digium.com/pub/zaptel/zaptel-1.4.1.tar.gz \ tar -xzf zaptel-1.4.1.tar.gz \ cd /usr/src/zaptel-1.4.1/ \ make clean ./configure make make install make config \ modprobe ztdummy Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] ipv6 patch
Is it exists? Regards, Hong Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097 ___ --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] chan-capi-HEAD and Asterisk 1.4.2
On Tue, 3 Apr 2007, Peer Oliver Schmidt wrote: Good Morning Armin, tried the patch, but it did not work. It waits quite a long time before the chan-capi error message comes up, according to the time stamp it is about 12 seconds. It is kind of strange, that the whole startup process for asterisk usually takes only about 4-5 seconds. That's too long, normaly the confirmation message arrives within a few msecs. So it seems that the driver isn't responding. Do you need additional information? Which card/driver do you use? [EMAIL PROTECTED]:~# lspci -s 0:0e -v 00:0e.0 Network controller: AVM Audiovisuelles MKTG Computer System GmbH A1 ISDN [Fritz] (rev 02) Subsystem: AVM Audiovisuelles MKTG Computer System GmbH FRITZ!Card ISDN Controller Flags: medium devsel, IRQ 10 Memory at ff001400 (32-bit, non-prefetchable) [size=32] I/O ports at dcc0 [size=32] [EMAIL PROTECTED]:~# capiinit status 1 fcpci running fcpci-dcc0-10A1 3.11-07 0xdcc0 10 Driver from ubuntu edgy I cannot tell anything about the AVM drivers. A debug log (capi trace) from the driver or kernelcapi helps to see what messages are wrong/missing. What is the best way to produce this? If the AVM driver can do that, I don't know. But on load of the module 'kernelcapi', you can specify the module parameter showcapimsgs=X where X is the verbose level. By default it is 0, which means no messges. You should set it to 3 to get the CAPI control messages on the kernel-console (logfile). Or even to 7 to have all CAPI messages (including data messages) which might be too much. Armin ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Hello Armin, here are the results, after modprobe kernelcapi showcapimsgs=3 /var/log/kern.log Apr 3 15:06:14 server42 kernel: [255113.053170] CAPI Subsystem Rev 1.1.2.8 Apr 3 15:06:18 server42 kernel: [255116.814334] fcpci: AVM FRITZ!Card PCI driver, revision 0.7.2 Apr 3 15:06:18 server42 kernel: [255116.814356] fcpci: (fcpci built on Feb 27 2007 at 21:22:25) Apr 3 15:06:18 server42 kernel: [255116.814367] fcpci: -- 32 bit CAPI driver -- Apr 3 15:06:18 server42 kernel: [255116.817598] PCI: Found IRQ 10 for device :00:0e.0 Apr 3 15:06:18 server42 kernel: [255116.817642] fcpci: AVM FRITZ!Card PCI found: port 0xdcc0, irq 10 Apr 3 15:06:18 server42 kernel: [255116.817653] fcpci: Loading... Apr 3 15:06:18 server42 kernel: [255116.817665] fcpci: Driver 'fcpci' attached to fcpci-stack. (152) Apr 3 15:06:18 server42 kernel: [255117.049308] fcpci: Stack version 3.11-07 Apr 3 15:06:18 server42 kernel: [255117.050442] kcapi: Controller 1: fcpci-dcc0-10 attached Apr 3 15:06:18 server42 kernel: [255117.050456] kcapi: card 1 fcpci-dcc0-10 ready. Apr 3 15:06:18 server42 kernel: [255117.050480] kcapi: notify up contr 1 Apr 3 15:06:19 server42 kernel: [255117.051283] fcpci: Loaded. Apr 3 15:06:22 server42 kernel: [255120.102281] capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Apr 3 15:06:35 server42 kernel: [255133.725987] kcapi: appl 1 up Apr 3 15:06:35 server42 kernel: [255133.727235] kcapi: put [0x1] id#1 FACILITY_REQ len=18 Apr 3 15:06:35 server42 kernel: [255133.727712] kcapi: got [0x1] id#1 FACILITY_CONF len=26 Apr 3 15:06:35 server42 kernel: [255133.729933] kcapi: appl 1 down Apr 3 15:06:35 server42 kernel: [255133.730585] kcapi: appl 1 up Apr 3 15:06:35 server42 kernel: [255133.731478] kcapi: put [0x1] id#1 LISTEN_REQ len=26 Apr 3 15:06:52 server42 kernel: [255150.690252] kcapi: appl 1 down At 15:06:35 the loading stopped. Is this helpful, or do you need a higher verbosity? -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Asterisk USER PORTAL
I'm trying to get the Mexuar development team to write some code to work with an existing asterisk USER PORTAL that presents a user with customized image of their Asterisk activities; * Address book, * Fop or some other kind of gui activity display * Voicemail access * Any other feature that should be integrated into a user portal page (maybe right click to dial or similar) We would like to develop some code that works with your existing User Portal to implement the Mexuar Corraleta IAX2 java applet softphone If you have a suitable portal or configuration or if you know of one I should be looking at can you please call me here in New York or email me some screen prints, if your portal is selected then this will give you additional functionality that you can market to your customers (we'll also throw in a license or two for you to set up as demo's for your clients). We basically just want to demonstrate this as a possible use for the Mexuar Corraleta technology on the demo pages of our website. URL links for you to check out; www.Mexuar.com http://www.mexuar.com/ www.Mexuar.com/Demo/Demo1 Flash Demo Page; http://www.mexuar.com/downloads/Level1Products/CorraletaDemoSound.swf Technical page for Asterisk System Integrators; http://www.voip-info.org/wiki/view/mexuar Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 Ph http://click.mexuar.com/webuser/click/7/userurl/Cognation http://click.mexuar.com/webuser/nojs/7/userurl/Cognation www.Mexuar.com http://www.mexuar.com/ Want to voice enable your website? Use Corraleta to reach your customers in 10 seconds or less. image001.gif Description: image001.gif ___ --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] Re: asterisk-users Digest, Vol 33, Issue 10
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] kirk Wireless + Asterisk 1.2.7 Mysql Realtime problem
Hello, We have 32 DECT clients connected to a Kirk Wireless 600/v3, the Kirk server is connected to an Asterisk 1.2.17 with realtime configuration (MySQL). Our problem is that our Asterisk Server uses the latest inserted user to places calls each time a call is made. Exemple: we have 3 phones with number: 618, 670, 610. The number 610 is the latest inserted phone in the Asterisk server. If the user 618 calls the number 670 the user of the 670 phone will see the number 610 on the phone display. someone have a solution ? Thanks for your help, Vincent Renaville ___ --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] CDR and RADIUS (cdr_radius) - working
Hi, I needed my CDR's to be stored using a RADIUS server. I found cdr_radius in the src directory. Looked in /docs for how to install it and I got it to work. Just want to say thanks to those who helped write this. Has anybody else used this, any comments, cause I found nothing using google, even voip-info has nothing on this module? -- thanks, Yusuf ___ --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] kirk Wireless + Asterisk 1.2.7 Mysql Realtime problem
This looks like the same issue I have with Astra phones, see the thread Multi-line phones - Asterisk uses wrong callerid. I do not know of a resolution for this yet. regards, Drew Vincent renaville wrote: Hello, We have 32 DECT clients connected to a Kirk Wireless 600/v3, the Kirk server is connected to an Asterisk 1.2.17 with realtime configuration (MySQL). Our problem is that our Asterisk Server uses the latest inserted user to places calls each time a call is made. Exemple: we have 3 phones with number: 618, 670, 610. The number 610 is the latest inserted phone in the Asterisk server. If the user 618 calls the number 670 the user of the 670 phone will see the number 610 on the phone display. someone have a solution ? Thanks for your help, Vincent Renaville -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.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
[asterisk-users] realtime problem
Dear the following is the asterisk's dbase(Mysql5). if the extension =17171000 asterisk run appdata=22, but I prefer to run appdata=333. let me know how I can run the appdata=3 best Mani mysql select * from ext; ++-++--+--+---+ | id | context | exten | priority | app | appdata | ++-++--+--+---+ | 1 | DID | _1.|1 | Dial | 222 | | 2 | DID | _1717. |1 | Dial | 333 | | 3 | DID | _171. |1 | Dial | 111 | ++-++--+--+---+ Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --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] Adding DND to dialplan
Hello -
I've read Asterisk should be able to activate a do not disturb feature
to turn off the ringers on extensions. I checked the wiki and can't
find documentation for how to do it.
Here's my attempt, added to extensions.conf:
[dnd-on]
exten = _#78,1,Answer
exten = _#78,n,Wait(1)
exten = _#78,n,Macro(user-callerid,)
exten = _#78,n,Set(DB(DND/${CALLERID(number)})=YES)
exten = _#78,n,Playback(do-not-disturbactivated)
exten = _#78,n,Macro(hangupcall,)
[dnd-off]
exten = _#79,1,Answer
exten = _#79,n,Wait(1)
exten = _#79,n,Macro(user-callerid,)
exten = _#79,n,dbDel(DND/${CALLERID(number)})
exten = _#79,n,Playback(do-not-disturbde-activated)
exten = _#79,n,Macro(hangupcall,)
;further down
include = dnd-on
include = dnd-off
- - -
Monitoring asterisk from the CLI, when I dial #78 on an extension, I
just get a fast busy signal and this information is reported on the
CLI:
Apr 3 10:41:33 WARNING[30702]: app_macro.c:144 macro_exec: No such
context 'macro-user-callerid' for macro 'user-callerid'
Apr 3 10:41:33 WARNING[30702]: func_db.c:97 function_db_write: DB
requires an argument, DB(family/key)=value
Apr 3 10:41:33 WARNING[30702]: file.c:504 ast_openstream_full: File
do-not-disturb does not exist in any format
Apr 3 10:41:33 WARNING[30702]: file.c:816 ast_streamfile: Unable to
open do-not-disturb (format unknown): No such file or directory
Apr 3 10:41:33 WARNING[30702]: app_playback.c:106 playback_exec:
ast_streamfile failed on Zap/2-1 for do-not-disturbactivated
Apr 3 10:41:33 WARNING[30702]: file.c:504 ast_openstream_full: File
activated does not exist in any format
Apr 3 10:41:33 WARNING[30702]: file.c:816 ast_streamfile: Unable to
open activated (format unknown): No such file or directory
Apr 3 10:41:33 WARNING[30702]: app_playback.c:106 playback_exec:
ast_streamfile failed on Zap/2-1 for do-not-disturbactivated
Apr 3 10:41:33 WARNING[30702]: app_macro.c:144 macro_exec: No such
context 'macro-hangupcall' for macro 'hangupcall'
- - -
Any tips?
All I really want to do is turn off the ringers / do not ring
extenstions when I've activated DND. Right now I'm just using a hack
which is to shutdown asterisk altogether when I don't want the phones
to ring, which of course also prevents dialing out, it's a
sledgehammer approach and I'm looking for something more typical.
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[asterisk-users] Require only GSM Codec
Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev ___ --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] chan-capi-HEAD and Asterisk 1.4.2
On Tue, 3 Apr 2007, Peer Oliver Schmidt wrote: Hello Armin, here are the results, after modprobe kernelcapi showcapimsgs=3 /var/log/kern.log ... Apr 3 15:06:22 server42 kernel: [255120.102281] capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Apr 3 15:06:35 server42 kernel: [255133.725987] kcapi: appl 1 up Apr 3 15:06:35 server42 kernel: [255133.727235] kcapi: put [0x1] id#1 FACILITY_REQ len=18 Apr 3 15:06:35 server42 kernel: [255133.727712] kcapi: got [0x1] id#1 FACILITY_CONF len=26 Apr 3 15:06:35 server42 kernel: [255133.729933] kcapi: appl 1 down Apr 3 15:06:35 server42 kernel: [255133.730585] kcapi: appl 1 up Apr 3 15:06:35 server42 kernel: [255133.731478] kcapi: put [0x1] id#1 LISTEN_REQ len=26 Apr 3 15:06:52 server42 kernel: [255150.690252] kcapi: appl 1 down At 15:06:35 the loading stopped. Is this helpful, or do you need a higher verbosity? Well, it confirmes what I have expected, but I cannot tell why it happens. The driver doesn't respond to the LISTEN_REQ command, that's why chan-capi shows an error. So with this info, the driver is the problem. Can you please do the same with 'showcapimsgs=2'? It may give more info on the commands itself, maybe some parameters are wrong here. Armin ___ --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] Adding DND to dialplan
Brian McEntire wrote:
I've read Asterisk should be able to activate a do not disturb feature
to turn off the ringers on extensions. I checked the wiki and can't
find documentation for how to do it.
Here's my attempt, added to extensions.conf:
[dnd-on]
exten = _#78,1,Answer
exten = _#78,n,Wait(1)
exten = _#78,n,Macro(user-callerid,)
exten = _#78,n,Set(DB(DND/${CALLERID(number)})=YES)
exten = _#78,n,Playback(do-not-disturbactivated)
exten = _#78,n,Macro(hangupcall,)
[dnd-off]
exten = _#79,1,Answer
exten = _#79,n,Wait(1)
exten = _#79,n,Macro(user-callerid,)
exten = _#79,n,dbDel(DND/${CALLERID(number)})
exten = _#79,n,Playback(do-not-disturbde-activated)
exten = _#79,n,Macro(hangupcall,)
;further down
include = dnd-on
include = dnd-off
- - -
Monitoring asterisk from the CLI, when I dial #78 on an extension, I
just get a fast busy signal and this information is reported on the
CLI:
Apr 3 10:41:33 WARNING[30702]: app_macro.c:144 macro_exec: No such
context 'macro-user-callerid' for macro 'user-callerid'
Apr 3 10:41:33 WARNING[30702]: func_db.c:97 function_db_write: DB
requires an argument, DB(family/key)=value
Apr 3 10:41:33 WARNING[30702]: file.c:504 ast_openstream_full: File
do-not-disturb does not exist in any format
Apr 3 10:41:33 WARNING[30702]: file.c:816 ast_streamfile: Unable to
open do-not-disturb (format unknown): No such file or directory
Apr 3 10:41:33 WARNING[30702]: app_playback.c:106 playback_exec:
ast_streamfile failed on Zap/2-1 for do-not-disturbactivated
Apr 3 10:41:33 WARNING[30702]: file.c:504 ast_openstream_full: File
activated does not exist in any format
Apr 3 10:41:33 WARNING[30702]: file.c:816 ast_streamfile: Unable to
open activated (format unknown): No such file or directory
Apr 3 10:41:33 WARNING[30702]: app_playback.c:106 playback_exec:
ast_streamfile failed on Zap/2-1 for do-not-disturbactivated
Apr 3 10:41:33 WARNING[30702]: app_macro.c:144 macro_exec: No such
context 'macro-hangupcall' for macro 'hangupcall'
- - -
Any tips?
Read the warnings.
Apart from that, what you do is good.
Right before you Dial() to a user you need to check the value
you have stored in DB(DND/${EXTEN}) and if it's YES simply
do not Dial().
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
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Re: [asterisk-users] Adding DND to dialplan
Brian,
DND is not real hard. You basicly want to to note the extension is set to
DND and then when someone calls that extension you check for DND status and
if it is yes then you go on to voicemail instead of dial. It sounds like you
are miss understanding the dialplan and how to use it. In your sample, do
the macros user-callerid and hangupcall exist? Do the sound files you
specified exist in var/lib/asterisk/sounds? A simple DND would look like so:
exten = *73,1,Answer()
exten = *73,n,Wait(0.5)
exten = *73,n,Set(DB(${CALLERID(number)}/DND)=1)
exten = *73,n,Playback(do-not-disturb)
exten = *73,n,Playback(enabled)
exten = *73,n,Hangup()
and then
When someone calls say extension 1000 I would have a macro check for :
exten = s,n,Set(DNDStatus=$[${DB(1000/DND)} = 1]) = returns a 1 if
enabled or a 0
exten = s,n,GoToIf($[${DNDStatus} = 1]?DND)
exten = s,n(DND),Voicemail([EMAIL PROTECTED],u)
On 4/3/07, Brian McEntire [EMAIL PROTECTED] wrote:
Hello -
I've read Asterisk should be able to activate a do not disturb feature
to turn off the ringers on extensions. I checked the wiki and can't
find documentation for how to do it.
Here's my attempt, added to extensions.conf:
[dnd-on]
exten = _#78,1,Answer
exten = _#78,n,Wait(1)
exten = _#78,n,Macro(user-callerid,)
exten = _#78,n,Set(DB(DND/${CALLERID(number)})=YES)
exten = _#78,n,Playback(do-not-disturbactivated)
exten = _#78,n,Macro(hangupcall,)
[dnd-off]
exten = _#79,1,Answer
exten = _#79,n,Wait(1)
exten = _#79,n,Macro(user-callerid,)
exten = _#79,n,dbDel(DND/${CALLERID(number)})
exten = _#79,n,Playback(do-not-disturbde-activated)
exten = _#79,n,Macro(hangupcall,)
;further down
include = dnd-on
include = dnd-off
- - -
Monitoring asterisk from the CLI, when I dial #78 on an extension, I
just get a fast busy signal and this information is reported on the
CLI:
Apr 3 10:41:33 WARNING[30702]: app_macro.c:144 macro_exec: No such
context 'macro-user-callerid' for macro 'user-callerid'
Apr 3 10:41:33 WARNING[30702]: func_db.c:97 function_db_write: DB
requires an argument, DB(family/key)=value
Apr 3 10:41:33 WARNING[30702]: file.c:504 ast_openstream_full: File
do-not-disturb does not exist in any format
Apr 3 10:41:33 WARNING[30702]: file.c:816 ast_streamfile: Unable to
open do-not-disturb (format unknown): No such file or directory
Apr 3 10:41:33 WARNING[30702]: app_playback.c:106 playback_exec:
ast_streamfile failed on Zap/2-1 for do-not-disturbactivated
Apr 3 10:41:33 WARNING[30702]: file.c:504 ast_openstream_full: File
activated does not exist in any format
Apr 3 10:41:33 WARNING[30702]: file.c:816 ast_streamfile: Unable to
open activated (format unknown): No such file or directory
Apr 3 10:41:33 WARNING[30702]: app_playback.c:106 playback_exec:
ast_streamfile failed on Zap/2-1 for do-not-disturbactivated
Apr 3 10:41:33 WARNING[30702]: app_macro.c:144 macro_exec: No such
context 'macro-hangupcall' for macro 'hangupcall'
- - -
Any tips?
All I really want to do is turn off the ringers / do not ring
extenstions when I've activated DND. Right now I'm just using a hack
which is to shutdown asterisk altogether when I don't want the phones
to ring, which of course also prevents dialing out, it's a
sledgehammer approach and I'm looking for something more typical.
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--
Bruce Reeves
Nortex Networks
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RE: [asterisk-users] realtime problem
Try looking at this link: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf +sorting Bobby ___ --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] TE120P and Unknown Signalling Method
On 4/2/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: I have a brand new TE120P card that I have installed and asterisk is not starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown signalling method 'pri_cpe' Make sure you have libpri installed and that it is the right version for your version of asterisk. ___ --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] master.csv interpretation
Anyone know of any tools for interpreting master.csv call logs? (Excel is kind of basic) ___ --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] Hints not working using SVN-branch-1.4-r59289
Group I'm having trouble getting hints to work correctly using SVN-branch-1.4-r59289 I have hints working on several other systems but I must be missing something this time around. VoIPGW*CLI show hints -= Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 3 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 2 [EMAIL PROTECTED] : State:Unavailable Watchers 4 [EMAIL PROTECTED] : State:Unavailable Watchers 4 --- - 6 hints registered Here is the sip.conf [general] context=default ; Default context for incoming calls allowguest=no ; Allow or reject guest calls (default is yes) allowoverlap=no ; Disable overlap dialing support. (Default is yes) ;allowtransfer=no ; Disable all transfers (unless enabled in peers or users) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls subscribecontext = default ; Set a specific context for SUBSCRIBE requests notifyringing = yes ; Notify subscriptions on RINGING state (default: no) notifyhold = yes; Notify subscriptions on HOLD state (default: no) limitonpeers=yes allow=ulaw [21] ;Bill Salmons type=peer username=21 callerid=Bill Salmons 21 secret=21 host=dynamic context=default mailbox=21 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=30 [23] ;Teresa Trautman type=peer username=23 callerid=Teresa Trautman 23 secret=23 host=dynamic context=default mailbox=23 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [25] ;Bill Goldsmith type=peer username=25 callerid=Bill Goldsmith 25 secret=25 host=dynamic context=default mailbox=25 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [26] ;Joelle Harris type=peer username=26 callerid=Joelle Harris 26 secret=26 host=dynamic context=default mailbox=26 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [29] ;Amanda Anderson type=peer username=29 callerid=Amanda Anderson 29 secret=29 host=dynamic context=default mailbox=29 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [30] ;Joelle Harris type=peer username=30 callerid=Liz Williamson 30 secret=30 host=dynamic context=default mailbox=30 canreinvite=no nat=1 qualify=yes Subscribecontext=default notifyringing=yes call-limit=300 [ata] type=peer username=ata host=dynamic context=default secret=ata here is the extensions.conf [default] include = parkedcalls exten = 21,hint(SIP/21) exten = 21,1,answer exten = 21,n,dial(sip/21|30|kw) exten = 21,n,voicemail([EMAIL PROTECTED]|u) exten = 23,hint(sip/23) exten = 23,1,answer exten = 23,n,dial(sip/23|30|kw) exten = 23,n,voicemail([EMAIL PROTECTED]|u) exten = 25,hint(SIP/25) exten = 25,1,answer exten = 25,n,dial(sip/25|30|kw) exten = 25,n,voicemail([EMAIL PROTECTED]|u) exten = 26,hint(SIP/26) exten = 26,1,answer exten = 26,n,dial(sip/26|30|kw) exten = 26,n,voicemail([EMAIL PROTECTED]|u) exten = 29,hint(SIP/29) exten = 29,1,answer exten = 29,n,dial(sip/29|30|kw) exten = 29,n,voicemail([EMAIL PROTECTED]|u) exten = 30,hint(SIP/30) exten = 30,1,answer exten = 30,n,dial(sip/30|30|kw) exten = 30,n,voicemail([EMAIL PROTECTED]|u) ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Hello Armin, thanks a lot for your help. Can you please do the same with 'showcapimsgs=2'? It may give more info on the commands itself, maybe some parameters are wrong here. Here you go. 17:23:17 is the magic time. Apr 3 17:23:09 server42 kernel: [263323.308388] fcpci: AVM FRITZ!Card PCI driver, revision 0.7.2 Apr 3 17:23:09 server42 kernel: [263323.308411] fcpci: (fcpci built on Feb 27 2007 at 21:22:25) Apr 3 17:23:09 server42 kernel: [263323.308421] fcpci: -- 32 bit CAPI driver -- Apr 3 17:23:10 server42 kernel: [263323.311559] PCI: Found IRQ 10 for device :00:0e.0 Apr 3 17:23:10 server42 kernel: [263323.311602] fcpci: AVM FRITZ!Card PCI found: port 0xdcc0, irq 10 Apr 3 17:23:10 server42 kernel: [263323.311613] fcpci: Loading... Apr 3 17:23:10 server42 kernel: [263323.311625] fcpci: Driver 'fcpci' attached to fcpci-stack. (152) Apr 3 17:23:10 server42 kernel: [263323.539987] fcpci: Stack version 3.11-07 Apr 3 17:23:10 server42 kernel: [263323.541140] kcapi: Controller 1: fcpci-dcc0-10 attached Apr 3 17:23:10 server42 kernel: [263323.541154] kcapi: card 1 fcpci-dcc0-10 ready. Apr 3 17:23:10 server42 kernel: [263323.541833] fcpci: Loaded. Apr 3 17:23:12 server42 kernel: [263325.975634] capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Apr 3 17:23:17 server42 kernel: [263330.892916] kcapi: put [0x1] FACILITY_REQ ID=001 #0x0001 LEN=0018 Apr 3 17:23:17 server42 kernel: [263330.892926] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.892933] FacilitySelector = 0x3 Apr 3 17:23:17 server42 kernel: [263330.892939] FacilityRequestParameter= 00 00 00 Apr 3 17:23:17 server42 kernel: [263330.892946] Apr 3 17:23:17 server42 kernel: [263330.893153] kcapi: got [0x1] FACILITY_CONF ID=001 #0x0001 LEN=0026 Apr 3 17:23:17 server42 kernel: [263330.893163] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.893169] Info = 0x0 Apr 3 17:23:17 server42 kernel: [263330.893176] FacilitySelector = 0x3 Apr 3 17:23:17 server42 kernel: [263330.893182] FacilityConfirmationParameter = 00 00 06 00 00\37703 00 00 Apr 3 17:23:17 server42 kernel: [263330.893190] Apr 3 17:23:17 server42 kernel: [263330.900689] kcapi: put [0x1] LISTEN_REQ ID=001 #0x0002 LEN=0026 Apr 3 17:23:17 server42 kernel: [263330.900699] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.900706] InfoMask = 0x Apr 3 17:23:17 server42 kernel: [263330.900713] CIPmask = 0x1fff03ff Apr 3 17:23:17 server42 kernel: [263330.900720] CIPmask2 = 0x0 Apr 3 17:23:17 server42 kernel: [263330.900726] CallingPartyNumber = default Apr 3 17:23:17 server42 kernel: [263330.900733] CallingPartySubaddress = default Apr 3 17:23:17 server42 kernel: [263330.900739] -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Adding DND to dialplan
Bruce Reeves wrote:
exten = *73,1,Answer()
exten = *73,n,Wait(0.5)
exten = *73,n,Set(DB(${CALLERID(number)}/DND)=1)
Would prefer Set(DB(${DND/CALLERID(num)})=1)
exten = *73,n,Playback(do-not-disturb)
exten = *73,n,Playback(enabled)
exten = *73,n,Hangup()
and then
When someone calls say extension 1000 I would have a macro check for :
exten = s,n,Set(DNDStatus=$[${DB(1000/DND)} = 1]) = returns a 1 if
enabled or a 0
exten = s,n,GoToIf($[${DNDStatus} = 1]?DND)
exten = s,n(DND),Voicemail([EMAIL PROTECTED],u)
More complete:
[macro-check-dnd]
exten = s,n,Answer()
exten = s,n,Wait(1)
exten = s,n,Set(DNDStatus=$[${DB(DND/${ARG1})} = 1])
exten = s,n,GotoIf($[${DNDStatus} = 1]?DND)
exten = s,n,Dial(SIP/${ARG1})
exten = s,n,Hangup()
exten = s,n(DND),Voicemail([EMAIL PROTECTED],u)
exten = s,n,Hangup()
[default]
exten = _,1,Macro(check-dnd,${EXTEN})
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] TE120P and Unknown Signalling Method
William Moore wrote: On 4/2/07, Klaverstyn, David C [EMAIL PROTECTED] wrote: I have a brand new TE120P card that I have installed and asterisk is not starting as I am getting the error: ERROR[5054] chan_zap.c: Unknown signalling method 'pri_cpe' Make sure you have libpri installed and that it is the right version for your version of asterisk. Make sure you had libpri installed BEFORE YOU BUILT ASTERISK. Asterisk won't build support for PRI if it does not see libpri installed when you build it. Install libpri, then rebuild and install Asterisk. ___ --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] Problem with a Zap channel
Hello, I have got two zap channels configured in our asterisk server, one of them is connected to the PSTN directly and the other one is connected to a gsm track, only for mobile calls. Both of them are basic lines. I just connect an iax softphone (idefisk) to the asterisk PBX. If I make a mobile call using the zap channel connected to a gsm track, the mobile I phoned does not hear me nothing. But If the call is made using the zap channel directly connected to the PSTN, both end points hear perfectly. Why this is happening? How could I solve that? Any clue will we wellcomed. Thanks in advance. VoipCrazy ___ --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] RE: Asterisk-Addon-1.4.0 MySQL module
I still can't figure out why res_config_mysql module not showing up with many attempt. Anyone have any idea on this? checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... yes configure: creating ./config.status config.status: creating build_tools/menuselect-deps config.status: creating makeopts Sincerely, K -Original Message- From: KC [mailto:[EMAIL PROTECTED] Sent: Friday, March 30, 2007 1:43 AM To: 'asterisk-users@lists.digium.com' Subject: Asterisk-Addon-1.4.0 MySQL I can't find anything about Asteirsk-Addon-1.4 MYSQL problem from googling. I thought it would be my error but surely not just tried asterisk 1.2.17 with addon 1.2.5 and it work. Does anyone else having problem to make res_config_mysql, cdr_addon_mysql and app_addon_sql_mysql in addon-1.4? Thanks for sharing There are no res_config_mysql and cdr_addon_mysql module after. /configure make all make install in asterisk module directory. It would be great if someone can give me some hint. I never experienced this before with 1.2 releases. Is there something changed on 1.4 releases? Or am I missing something. I am about to pull my hair out after many hours looking at the monitor. uname -a Linux 2.6.20-1.2933.fc6 #1 SMP Mon Mar 19 10:42:48 EDT 2007 i686 i686 i386 GNU/Linux rpm -qa | grep -i mysql mysql-5.0.27-1.fc6 php-mysql-5.1.6-3.4.fc6 mysql-devel-5.0.27-1.fc6 perl-DBD-MySQL-3.0007-1.fc6 mysql-server-5.0.27-1.fc6 *CLI core show version Asterisk 1.4.2 on a i686 running Linux on 2007-03-28 05:45:27 UTC *CLI show modules like mysql Module Description Use Count 0 modules loaded Thank You K -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.22/739 - Release Date: 3/29/2007 1:36 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.22/739 - Release Date: 3/29/2007 1:36 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.25/745 - Release Date: 4/3/2007 12:48 PM ___ --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] Require only GSM Codec
On Tue, 3 Apr 2007, Sanjay Rajdev wrote: Hello All, I would like to only use the gsm codec for all the calls, is it possible Yes, it's possible. I want to use minimum possible bandwidth as we have most of calls over Internet. Good move if you're prepared to sacrifice call quality, however not all devices support GSM I guess what you're actually asking is how to do it? Get the book - http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 it's free, or buy a paper copy from Amazon. But to start, you need to look up the sip.conf and iax.conf files. Put this in them at the appropriate point disallow=all allow=gsm Gordon ___ --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] Require only GSM Codec
Hi Sanjay, I'm not sure about that, but I think you can configure it in, for example, /etc/asterisk/sip.conf. There is an option that you configure for each channel like: only=gsm It instructs the sip protocol, that only gsm codec must be used. I hope it has helped you. Regards, Ronaldo. On 4/3/07, Sanjay Rajdev [EMAIL PROTECTED] wrote: Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems with TE110P
On Tue, Apr 03, 2007 at 10:18:56AM +1000, Klaverstyn, David C wrote: OK, Found the problem. It looks like the configuration file is not correct. I added the following line to /etc/sysconfig/zaptel MODULES=$MODULES wcte12xp # TE120P - Single Span T1 Card Actually, make that: MODULES=wcte12xp a single line. Nothing more is needed. Once I did this all is now working. Editing the zaptel.sysconfig file in the zaptel source code will also do the same. So I'm guessing anyone with a TE120P card will need to do the same until Asterisk update the files for the TE120P What? ship a modprobe file that probes for every card module even if you just need ztdummy? Fun indeed. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Re: asterisk-users Digest, Vol 33, Issue 11
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] SIP 484 (Early Dial) and International Dialing
On 4/3/07, Olle E Johansson [EMAIL PROTECTED] wrote: I turned on the early dial option on the GXP, which causes each digit to be sent as it is pressed, and the user response was much more favourable. Now I come to set up my international dialplans and I'm running into a problem. You are assuming that SIP works like zaptel in the dialplan, but it does not. You propably need to re-configure your phones. I recognize the difference between Zaptel and SIP. The option to turn on 'early dial' in the GXP2000 makes the user experience of dialing on a SIP phone more like a Zaptel (or traditional analong phone). SIP phones have many similarities to cell phones, and while my users are familiar with the enter entire number then press send concept, they don't seem to intuitively apply it to their desk phones. The process flow for the GXP2000 out of the box is: 1. user starts dialing 2. GXP collects digits until the keypad timeout (default 4 seconds) expires or the # button is pressed 3. GXP sends entire dialstring as an INVITE, to get back something from *, typically a 404 or 100 If you turn on early dialing (against * v1.2), then the process changes to 1. user starts dialing 2. digit is sent as an invite, GXP waits for 100/404/484 3. If 484 is received, GXP waits for next digit from the user. When received, both digits are sent as an invite, and the wait for 100/404/484 starts again 4. If the keypad timeout expires, the GXP looks to see if the last received message was a 484. If so, the user gets a congestion tone The problem comes in with users who are used to dialing 9 and waiting for a redial tone. With early dial turned off, they hit 9, wait, don't get a redial tone because there's no such thing in *. If they start dialing the rest of their number before the keypad timeout expires, then all is good. If they don't, then the GXP sends an invite for '9', which get rejected with a 404 or possibly handled using Playback(pbx-invalid) or something similar. You can avoid the potential for the timeout to expire by increasing it from the default of 4 seconds to 5 or 6, but then you force users to press # or wait for the initial INVITE to be sent when they dial an extension correctly. I've found this dead time to be a source of confusion and frustration for users. After all, if I know my buddy is at extension 301, shouldn't the call go through the second I finish dialing the 1 key? The early dial mode in the GXP allows you to set the timeout value higher (to avoid sending the incomplete '9' invite) without the downside of forcing correctly dialed extensions to timeout before the invite is sent. It just makes the phone work more like the way people are used to phones working. Unfortunately, it has this downside when the number of digits in the extension is variable, and I didn't account for this during planning. I am thus left with a choice between preserving the traditional user experience and letting calls to variable-length extensions work properly. Or changing phones - obviously something with an on-phone dialplan would not suffer from this problem, nor would it need something like early dial. The DISA() solution works. Once it's invoked, the user experience is just like a Zap channel. I was just hoping that there might be some tips and tricks type of workaround for this situation. If not, DISA() will service my needs for now. -- j. ___ --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] RE: Asterisk-Addon-1.4.0 MySQL module
KC wrote: I still can't figure out why res_config_mysql module not showing up with many attempt. Anyone have any idea on this? checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... yes configure: creating ./config.status config.status: creating build_tools/menuselect-deps config.status: creating makeopts Sincerely, K -Original Message- From: KC [mailto:[EMAIL PROTECTED] Sent: Friday, March 30, 2007 1:43 AM To: 'asterisk-users@lists.digium.com' Subject: Asterisk-Addon-1.4.0 MySQL I can't find anything about Asteirsk-Addon-1.4 MYSQL problem from googling. I thought it would be my error but surely not just tried asterisk 1.2.17 with addon 1.2.5 and it work. Does anyone else having problem to make res_config_mysql, cdr_addon_mysql and app_addon_sql_mysql in addon-1.4? Thanks for sharing There are no res_config_mysql and cdr_addon_mysql module after. /configure make all make install in asterisk module directory. It would be great if someone can give me some hint. I never experienced this before with 1.2 releases. Is there something changed on 1.4 releases? Or am I missing something. I am about to pull my hair out after many hours looking at the monitor. uname -a Linux 2.6.20-1.2933.fc6 #1 SMP Mon Mar 19 10:42:48 EDT 2007 i686 i686 i386 GNU/Linux rpm -qa | grep -i mysql mysql-5.0.27-1.fc6 php-mysql-5.1.6-3.4.fc6 mysql-devel-5.0.27-1.fc6 perl-DBD-MySQL-3.0007-1.fc6 mysql-server-5.0.27-1.fc6 *CLI core show version Asterisk 1.4.2 on a i686 running Linux on 2007-03-28 05:45:27 UTC Did you install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf libtool make automake automake14 automake15 automake16 automake17 bison byacc flex libtermcap libtermcap-devel newt newt-devel ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel mysqlclient10 mysqlclient10-devel mysqlclient12 mysqlclient12-devel mysqlclient14 mysqlclient14-devel ? Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] misdn and debian
On Tue, Apr 03, 2007 at 11:15:36AM +0200, Christoph Fürstaller wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Have you got zaptel installed on your box? And loading the zaphfc module during bootup? I discovered the problem you reported when I switched to mISDN and having installed/loading zaphfc during bootup. Than asterisk doesn't start and the system hangs. zaphfc? How do you load it automatically? Unlike most other modules it doesn't get hotplugged in, so you have to set to to explicitly load. Not to mention that it is for HFC-s cards and not for the AVM one. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] misdn and debian
On Mon, Apr 02, 2007 at 08:30:57PM +0300, Giedrius Augys wrote: Hi, I have FRITZ!Card PCI card. I have installed misdn-1.1.0 on stable debian 3.1 with 2.6.8. kernel. Then I reboot system, and it doesn't boot, it stops near Apache2 starting I started my system with recovery kernel, and tun off misd, then my system works fine. I think it's problem with memory. Have you tried memtest? apt-get install memtest86 , enable it in /etc/grub/menu.lst and run 'update-grub' . Has anybody debian and misdn working fine? Maybe you can advices , what kernel and misdn versions to use... Let's think: what comes shortly after apache? maybe asterisk? to get a better idea: ls /etc/rc2.d This ialso suggests that you use asterisk from your own build rather than from the package. In the package asterisk starts after most other services, in order for the service asterisk to start after the service zaptel. Is asterisk running with the option '-p'? If so: disable it for the purpose of testing. It makes an asterisk 100% CPU loop into a hanged system. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Zaptel 1.4.1 Install Modules CentOS
On Tue, Apr 03, 2007 at 11:57:57AM +0100, Chris Blunt wrote: Hi All, I have a CentOS server that I am trying to configure Asterisk on 1.4 on. Everything seems to go ok, with regards to compiling Zaptel, Libpri, Asterisk (will be using kernel 2.6 timer and ztdummy) Unfortunately I can't insmod / modprobe ztdummy. Have you run 'make install'? What is the output of modinfo zaptel Any change if you run: depmod [root @xyz src]# modprobe ztdummy FATAL: Module ztdummy not found. FATAL: Error running install command for ztdummy [EMAIL PROTECTED] src]# insmod ztdummy insmod: can't read 'ztdummy': No such file or directory insmod ./ztdummy.ko But it should fail (e.g: because zaptel is not loaded). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] DTMF problem with 1.4.1
it shows empty string On 4/3/07, Rizwan Hisham [EMAIL PROTECTED] wrote: You can use the following to display what you receive from user (dtmf): exten= 1,1,Read(test) exten= 1,2,NoOp(DTMF Received: $test) exten= 1,3,Hangup On 4/3/07, Nitin Gupta [EMAIL PROTECTED] wrote: I upgraded to 1.4.1 and my DTMF has stopped working, I tried rfc2833compensate=yes and relaxdtmf=yes etc but none working. Everything seems to work fine with 1.2.10 Is there any way I dump the dtmf data packets received by asterisk on console? Any idea or pointers to debug the issue will be much appreciated. thanks, Nitin ___ --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 -- Regards Rizwan Hisham Software Engineer ___ --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] Interconnecting Cisco 1760 routers with Asterisk
Good day everyone. I have Cisco 1760 routers that do site to site voip. Each router has 2 fxs ports that connect to the local pbx and use sip to connect to other routers over the WAN. I am thinking of putting in an asterisk box at the hub site for interconnectivity with our global office voip provider. This provider runs asterisk. Question is - can Cisco 1760 routers make/receive calls to/fro asterisk? if yes, any sample configuration please? Thanks and regards Joesph Abuja, Nigeria ___ --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] DTMF problem with 1.4.1
That would be because $test is not a valid dialplan variable. You would
want ${test}
Nitin Gupta wrote:
it shows empty string
On 4/3/07, *Rizwan Hisham* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
You can use the following to display what you receive from user (dtmf):
exten= 1,1,Read(test)
exten= 1,2,NoOp(DTMF Received: $test)
exten= 1,3,Hangup
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Re: [asterisk-users] understanding what h extension does [ISSUE SOLVED]
On Tuesday 03 April 2007 07:48, Alan Chandler wrote:
I am trying to make a dialplan that when I dial 90 I can go round a
whole set of extensions and leave them a short message, hangup and go
on the next one.
I use the M facility of dial, with something like this
[messages]
exten = 90,n(calcnextchan),Set(DIALCHAN=...)
exten = 90,n,Dial(${DIALCHAN},30,M(domessage))
exten = 90,n,Goto(calcnextchan)
[macro-domessage]
exten = s,1,Playback(message)
exten = s,n,Set(MACRO_RESULT=CONTINUE)
[There is actually more logic to check for busy dial channels and
retry them later]
This seems to work fine until one of the callees hangs up before the
message is played. at which point my call is terminated.
OK, its a logic problem. If the caller hangs up before playback is
complete MACRO_RESULT has not been set, so the call is bridged and then
hung up. If I set MACRO_RESULT as the first action of the call macro,
then any interruption from the far end hanging up means that the
dialplan just continues without the call having been bridged.
--
Alan Chandler
http://www.chandlerfamily.org.uk
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[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] Adding DND to dialplan
Ahh... got it now. Thanks for all the replies.
I was thinking that it was a function that was already built in, but
I see by setting a value and then testing it before ringing
extensions, it's easily added to the dialplan.
On 4/3/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Bruce Reeves wrote:
exten = *73,1,Answer()
exten = *73,n,Wait(0.5)
exten = *73,n,Set(DB(${CALLERID(number)}/DND)=1)
Would prefer Set(DB(${DND/CALLERID(num)})=1)
exten = *73,n,Playback(do-not-disturb)
exten = *73,n,Playback(enabled)
exten = *73,n,Hangup()
and then
When someone calls say extension 1000 I would have a macro check for :
exten = s,n,Set(DNDStatus=$[${DB(1000/DND)} = 1]) = returns a 1 if
enabled or a 0
exten = s,n,GoToIf($[${DNDStatus} = 1]?DND)
exten = s,n(DND),Voicemail([EMAIL PROTECTED],u)
More complete:
[macro-check-dnd]
exten = s,n,Answer()
exten = s,n,Wait(1)
exten = s,n,Set(DNDStatus=$[${DB(DND/${ARG1})} = 1])
exten = s,n,GotoIf($[${DNDStatus} = 1]?DND)
exten = s,n,Dial(SIP/${ARG1})
exten = s,n,Hangup()
exten = s,n(DND),Voicemail([EMAIL PROTECTED],u)
exten = s,n,Hangup()
[default]
exten = _,1,Macro(check-dnd,${EXTEN})
Regards,
Philipp
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create new ones.
Asterisk? - http://www.das-asterisk-buch.de
Geschäftsführer: Stefan Wintermeyer
Handelsregister: Neuwied B 14998
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Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12
Ok, I'll bite. This is the 4th message like this I've gotten today. I don't speak French but it looks like an autoresponder. If so, why is it replying back to the list, why not on every message sent, and why is it incrementing the issue number? Or am I missing something? Jay [EMAIL PROTECTED] wrote: Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12
I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 4/3/2007 2:29:11 PM Ok, I'll bite. This is the 4th message like this I've gotten today. I don't speak French but it looks like an autoresponder. If so, why is it replying back to the list, why not on every message sent, and why is it incrementing the issue number? Or am I missing something? Jay [EMAIL PROTECTED] wrote: Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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 - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --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] Re: asterisk-users Digest, Vol 33, Issue 12
Jay Moore wrote: Ok, I'll bite. This is the 4th message like this I've gotten today. I don't speak French but it looks like an autoresponder. I'm away from ... to ... I'm going to respond to your message when I'm back. In urgent cases contact ... or ... If so, why is it replying back to the list, I thought mailman would catch autoresponders(?) why not on every message sent, Probably beacause he subscribed to the digest, not to the real list. and why is it incrementing the issue number? No idea. Or am I missing something? The digest. ;) /kick him ;) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Re: asterisk-users Digest, Vol 33, Issue 12
Unless he's European, if so these messages will stop next Wednesday, hopefully Bails john beaman wrote: I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 4/3/2007 2:29:11 PM Ok, I'll bite. This is the 4th message like this I've gotten today. I don't speak French but it looks like an autoresponder. If so, why is it replying back to the list, why not on every message sent, and why is it incrementing the issue number? Or am I missing something? Jay [EMAIL PROTECTED] wrote: Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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 - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12
Philipp Kempgen wrote: /kick him ;) I'm not really sure whether fb is to blame or if mailman should be in charge of filtering autoresponders. Thus I did not put fb on my black(mail)list - yet. :) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Re: asterisk-users Digest, Vol 33, Issue 12
john beaman wrote: I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... This is from April 2nd to April 11th. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] Re: asterisk-users Digest, Vol 33, Issue 12
john beaman wrote: I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... No. 11/04 = the 11th of April You US guys mix that up. ;) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] Re: asterisk-users Digest, Vol 33, Issue12
John, it's the 11th of April not 4th of November. I think everyone on this list should send [EMAIL PROTECTED] one of their favorite photos, nothing rude or crass, just a nice thank you for wasting our bandwidth. Regards, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of john beaman Sent: Tuesday, 3 April 2007 3:43 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue12 I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 [EMAIL PROTECTED] 4/3/2007 2:29:11 PM Ok, I'll bite. This is the 4th message like this I've gotten today. I don't speak French but it looks like an autoresponder. If so, why is it replying back to the list, why not on every message sent, and why is it incrementing the issue number? Or am I missing something? Jay [EMAIL PROTECTED] wrote: Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue12
November? It's DD/MM/ in his case, not MM/DD/. Either way, even two days is more than enough for me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of john beaman Sent: Tuesday, April 03, 2007 12:43 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue12 I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... John Beaman Telecom Specialist Voice Telecommunications Services Department. Good Samaritan National Campus 605-362-3331 ___ --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] chan-capi-HEAD and Asterisk 1.4.2
On Tue, 3 Apr 2007, Peer Oliver Schmidt wrote: Hello Armin, thanks a lot for your help. Can you please do the same with 'showcapimsgs=2'? It may give more info on the commands itself, maybe some parameters are wrong here. Here you go. 17:23:17 is the magic time. This log below shows no error in parameters, but the problem is still the same: the fcpci driver doesn't respond and I cannot tell why. Armin Apr 3 17:23:09 server42 kernel: [263323.308388] fcpci: AVM FRITZ!Card PCI driver, revision 0.7.2 Apr 3 17:23:09 server42 kernel: [263323.308411] fcpci: (fcpci built on Feb 27 2007 at 21:22:25) Apr 3 17:23:09 server42 kernel: [263323.308421] fcpci: -- 32 bit CAPI driver -- Apr 3 17:23:10 server42 kernel: [263323.311559] PCI: Found IRQ 10 for device :00:0e.0 Apr 3 17:23:10 server42 kernel: [263323.311602] fcpci: AVM FRITZ!Card PCI found: port 0xdcc0, irq 10 Apr 3 17:23:10 server42 kernel: [263323.311613] fcpci: Loading... Apr 3 17:23:10 server42 kernel: [263323.311625] fcpci: Driver 'fcpci' attached to fcpci-stack. (152) Apr 3 17:23:10 server42 kernel: [263323.539987] fcpci: Stack version 3.11-07 Apr 3 17:23:10 server42 kernel: [263323.541140] kcapi: Controller 1: fcpci-dcc0-10 attached Apr 3 17:23:10 server42 kernel: [263323.541154] kcapi: card 1 fcpci-dcc0-10 ready. Apr 3 17:23:10 server42 kernel: [263323.541833] fcpci: Loaded. Apr 3 17:23:12 server42 kernel: [263325.975634] capi20: Rev 1.1.2.7: started up with major 68 (middleware+capifs) Apr 3 17:23:17 server42 kernel: [263330.892916] kcapi: put [0x1] FACILITY_REQ ID=001 #0x0001 LEN=0018 Apr 3 17:23:17 server42 kernel: [263330.892926] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.892933] FacilitySelector = 0x3 Apr 3 17:23:17 server42 kernel: [263330.892939] FacilityRequestParameter = 00 00 00 Apr 3 17:23:17 server42 kernel: [263330.892946] Apr 3 17:23:17 server42 kernel: [263330.893153] kcapi: got [0x1] FACILITY_CONF ID=001 #0x0001 LEN=0026 Apr 3 17:23:17 server42 kernel: [263330.893163] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.893169] Info= 0x0 Apr 3 17:23:17 server42 kernel: [263330.893176] FacilitySelector = 0x3 Apr 3 17:23:17 server42 kernel: [263330.893182] FacilityConfirmationParameter = 00 00 06 00 00\37703 00 00 Apr 3 17:23:17 server42 kernel: [263330.893190] Apr 3 17:23:17 server42 kernel: [263330.900689] kcapi: put [0x1] LISTEN_REQ ID=001 #0x0002 LEN=0026 Apr 3 17:23:17 server42 kernel: [263330.900699] Controller/PLCI/NCCI = 0x1 Apr 3 17:23:17 server42 kernel: [263330.900706] InfoMask= 0x Apr 3 17:23:17 server42 kernel: [263330.900713] CIPmask= 0x1fff03ff Apr 3 17:23:17 server42 kernel: [263330.900720] CIPmask2= 0x0 Apr 3 17:23:17 server42 kernel: [263330.900726] CallingPartyNumber = default Apr 3 17:23:17 server42 kernel: [263330.900733] CallingPartySubaddress = default Apr 3 17:23:17 server42 kernel: [263330.900739] -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Play blank sound while VM recording?
Greetings, (Apologies if this is an FAQ, but I've Googled for hours and haven't come up with anything yet.) I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any RTP packets back through the trunk after the beep is played. This is fine and probably should be the expected behavior, except that after 30 seconds to a minute of not seeing any RTP traffic coming from the PBX, the trunk appears to make the faulty assumption that the PBX is gone and hangs up the call. I've called the trunk provider and they said two things. 1) This is indeed what their trunk was programmed to do. 2) No, they won't change it. We're working on switching the customer to a trunk provider with a bit more clue, but in the meantime, how can I have Asterisk play an empty sound file while the caller is leaving a voicemail message just to keep the RTP traffic flowing? This installation of Asterisk was designed by someone else and I have limited personal experience with Asterisk configuration files, so an example would be appreciated if possible. Thanks! -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.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
[asterisk-users] asterisk and mplayer
Odd question here but if I have asterisk running on PC (and mplayer installed). and a video phone calls up the asterisk PC can that video image be played on mplayer? If so how do I do that? How can asterisk pipe the video into mplayer so as to display the video image on screen? Thanks, Jerry ___ --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] Re: asterisk-users Digest, Vol 33, Issue 12
john beaman wrote: I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... In civilised countries, we read that as 2nd April 2007 to the 11th April 2007! regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation 416-593-6767 x322 www.oanda.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
Re: [asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 12
Ah, yes. One of the many differences between the US and the rest of the world. [EMAIL PROTECTED] 4/3/2007 2:52:16 PM john beaman wrote: I too was curious about this, so I copied the text into Babel Fish, and this is the result: I miss of the 2/04/2007 to the 11/04/2007. I will answer your message as of my return. For any urgency, to contact Emmanuelle Parache Moga or Cédric Buzay. If this guy is really going to be out until November these messages will get rather tiresome... This is from April 2nd to April 11th. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 - This email transmission and any documents, files or previous email messages attached to it may contain information that is confidential or legally privileged. If you are not the intended recipient, you are hereby notified that any disclosure, copying, printing, distributing or use of this transmission is strictly prohibited. If you have received this transmission in error, please immediately notify the sender by telephone or return email and delete the original transmission and its attachments without reading or saving in any manner. The Evangelical Lutheran Good Samaritan Society. - ___ --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] T400P 4 Port T1 Cards for Sale
Hello Everyone, I have two never-used, still in the static-bag T400P Cards that I bought a long while back that I'd like to get rid of. Before I ever got a chance to use them, I bought Sangoma A400 Cards instead, and now I definately don't need them any longer. They have the Dallas DS21Q352 Chip on them (A4 Revision). I'll be happy to test them before I ship them if you like. Anyone interested, send me an offer. Reasonable offers accepted, but I don't 'need' to get rid of them, so please don't low-ball them too badly. I'll sell them seperately or together. Please send offers to me directly (enigma81 at rock dot com) John -- You Rock! Your E-Mail Should Too! Signup Now at Rock.com and get 250MB of Storage! http://webmail.rock.com/signup/ ___ --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] chan-capi-HEAD and Asterisk 1.4.2
Good evening Armin, This log below shows no error in parameters, but the problem is still the same: the fcpci driver doesn't respond and I cannot tell why. Ok. Thanks for your assistants anyhow. What strikes me as strange ist the fact, that turning on verbose helps to circumvent the problem. Thanks again, and have a good night. -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --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] Play blank sound while VM recording?
Charles Ulrich wrote: I have an Asterisk system deployed at a customer's site. It is connected to the outside world by a local SIP provider. When someone calls in through the trunk to leave a voicemail, Asterisk is not sending any RTP packets back through the trunk after the beep is played. This is fine and probably should be the expected behavior, except that after 30 seconds to a minute of not seeing any RTP traffic coming from the PBX, the trunk appears to make the faulty assumption that the PBX is gone and hangs up the call. Maybe this is what you need?: ;rtpkeepalive=secs; Send keepalives in the RTP stream to keep NAT open ; (default is off - zero) (in sip.conf, [general] section) Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] asterisk and mplayer
Jerry Geis wrote: Odd question here but if I have asterisk running on PC (and mplayer installed). and a video phone calls up the asterisk PC can that video image be played on mplayer? Afaik Asterisk does not have any support for video streams yet. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk? - http://www.das-asterisk-buch.de Geschäftsführer: Stefan Wintermeyer Handelsregister: Neuwied B 14998 ___ --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] adding chan_celliax
On Tue, Apr 03, 2007 at 08:50:02AM +0200, Giovanni Maruzzelli wrote: Ciao Patricio, have you compiled Asterisk from sources? At the moment you can only add chan_celliax support if you compiled from source. If this is the case, I can give you full instruction. And if from packages: http://updates.xorcom.com/contrib/celliax/ could be a good start. Though I only gotten it to build. Generally there shouldn't be a problem building chan_celliax vs. the package asterisk-dev . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] x100p not showing in core show channels
Hi, I recently decided to change my setup from AsteriskNow to plain-asterisk 1.4, which I wanted to set up and configure myself on a server running Debian Etch 64bit version. Hardware: Asrock motherboard, model 775Dual880-Pro, with a Celeron D running at 2.8GHz, 1GB memory, standard Nvidia GF4MX videocard, and one X100P clone card. Running the AsteriskNow, everything worked fine, except for incoming calls, not being routed right, but they entered the system, and mostly they ended up into the voicemail. To change that behaviour and to have more control of what I'm doing, I reinstalled the same machine with Debian Etch, the 64bit version, as the CPU (and the replacing one in a few weeks) runs EM64T nicely... The setup ran OK, compilation etc too, except for zttool which I still cannot compile. When configging the server, I used several HowTo's and guides, but no solution: hereby the config parts that I did/changed: /etc/zaptel.conf: # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # This file is parsed by the Zaptel Configurator, ztcfg # It must be in the module loading order # Span 1: WCFXO/0 Generic Clone Board 1 fxsks=1 # Global data loadzone= be defaultzone = be /etc/asterisk/zapata.conf: [channels] signalling=fxs_ks group=1 context=incoming channel=1 ;X100P /etc/asterisk/extensions.conf: [incoming] exten = s,1,Echo ;for testing the connection ;exten = s,1,Playback,demo-thanks ;for playing a file Nothing happens when dialing in. BUT: ztmonitor 01 -vv gives levels, and when dialing in, the levels change according to dialtone in my phone I use for calling the server. AND: core show channels gives me this: asterisk*CLI core show channels verbose Channel Context ExtensionPrio State Application Data CallerIDDuration Accountcode BridgedTo 0 active channels 0 active calls What am I doing wrong??? Anyone that can give a hand? Thanks! Just email me! bram_at_antwerpen_dot_be ___ --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] Lithuania
Hi All! Maybe a little of topic. Bout coming from Sweden and needing to call Lithuania a lot am I wondering if anyone on the list could recommend a sheep service in Lithuania to connect my Asterisk to. A local number are not necessary bout preferd for incoming calls for my contacts. Regards Mattias Andersson Adress: Mattias Andersson Storskiftesvägen 6 S-145 60 Norsborg Mobil: +46-70-799 44 41 Email: [EMAIL PROTECTED] Skype: eskes1 ___ --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] x100p not showing in core show channels
On Tue, Apr 03, 2007 at 11:17:57PM +0200, bram kortleven wrote: Hi, I recently decided to change my setup from AsteriskNow to plain-asterisk 1.4, which I wanted to set up and configure myself on a server running Debian Etch 64bit version. Hardware: Asrock motherboard, model 775Dual880-Pro, with a Celeron D running at 2.8GHz, 1GB memory, standard Nvidia GF4MX videocard, and one X100P clone card. Running the AsteriskNow, everything worked fine, except for incoming calls, not being routed right, but they entered the system, and mostly they ended up into the voicemail. To change that behaviour and to have more control of what I'm doing, I reinstalled the same machine with Debian Etch, the 64bit version, as the CPU (and the replacing one in a few weeks) runs EM64T nicely... The setup ran OK, compilation etc too, except for zttool which I still cannot compile. apt-get build-dep zaptel Or specifically: apt-get install libncurses-dev (or something similar) When configging the server, I used several HowTo's and guides, but no solution: hereby the config parts that I did/changed: /etc/zaptel.conf: # Autogenerated by /usr/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # This file is parsed by the Zaptel Configurator, ztcfg # It must be in the module loading order # Span 1: WCFXO/0 Generic Clone Board 1 fxsks=1 # Global data loadzone= be defaultzone = be /etc/asterisk/zapata.conf: [channels] signalling=fxs_ks group=1 context=incoming channel=1 ;X100P /etc/asterisk/extensions.conf: [incoming] exten = s,1,Echo ;for testing the connection ;exten = s,1,Playback,demo-thanks ;for playing a file Nothing happens when dialing in. BUT: ztmonitor 01 -vv gives levels, and when dialing in, the levels change according to dialtone in my phone I use for calling the server. AND: core show channels gives me this: asterisk*CLI core show channels verbose Channel Context ExtensionPrio State Application Data CallerIDDuration Accountcode BridgedTo 0 active channels 0 active calls What am I doing wrong??? Anyone that can give a hand? What is the output of: zap show channels -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Re: asterisk-users Digest, Vol 33, Issue12
Hello Darryl, * Darryl Dunkin [EMAIL PROTECTED] [03-04-07 12:56]: November? It's DD/MM/ in his case, not MM/DD/. Either way, even two days is more than enough for me. is the format not? MM/DD/ DD.MM. Best regards, Matthias -- Programming today is a race between software engineers striving to build bigger and better idiot-proof programs, and the universe trying to produce bigger and better idiots. So far, the universe is winning. -- Rich Cook ___ --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] stun
Is it possible to install a stun server on asterisk? joe a. ___ --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] DTMF via IAX ignored after a few seconds
I'm new to this list, and I apologize if this is an already answered question, but my Google-fu was not strong enough to find the answer if it was. I'm having a problem with DTMF on incoming IAX calls. For the first few seconds of the call (between maybe 1 and 15, it varies from call to call) everything works fine. After that I continue get DTMF_E messages from the remote IAX server and continue to send back ACKs, but the tones stop being processed. Everything seems to work fine from my internal SIP phones as well as on inbound calls if I switch inbound routing to SIP. I can make SIP work if I need to, but I'd like to use IAX for a number of reasons, and at the very least I'd like to understand the problem before I give up and switch. Here's an excerpt from my console log showing the working and non-working DTMF_E messages, which look identical to me. The complete log for the call follows for context. If there's some other bit of logging I could turn on that might show me what happens to the tones after they're acknowledged I'd be glad to know. Zach #=== -- Goto (cynic-closed,s,1) -- Executing [EMAIL PROTECTED]:1] BackGround(IAX2/vitel-inbound-1, normalized/technical-supportnormalized/is-curntly-unavail) in new stack -- IAX2/vitel-inbound-1 Playing 'normalized/technical-support' (language 'en') Rx-Frame Retry[ No] -- OSeqno: 003 ISeqno: 004 Type: DTMF_E Subclass: 4 Timestamp: 05403ms SCall: 00023 DCall: 1 [64.2.142.31:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 004 Type: IAX Subclass: ACK Timestamp: 05403ms SCall: 1 DCall: 00023 [64.2.142.31:4569] Rx-Frame Retry[ No] -- OSeqno: 004 ISeqno: 004 Type: DTMF_E Subclass: 1 Timestamp: 05623ms SCall: 00023 DCall: 1 [64.2.142.31:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 005 Type: IAX Subclass: ACK Timestamp: 05623ms SCall: 1 DCall: 00023 [64.2.142.31:4569] Rx-Frame Retry[ No] -- OSeqno: 005 ISeqno: 004 Type: DTMF_E Subclass: 1 Timestamp: 05783ms SCall: 00023 DCall: 1 [64.2.142.31:4569] Tx-Frame Retry[-01] -- OSeqno: 004 ISeqno: 006 Type: IAX Subclass: ACK Timestamp: 05783ms SCall: 1 DCall: 00023 [64.2.142.31:4569] #=== The tones for 411 above work fine and call is routed to the directory app == CDR updated on IAX2/vitel-inbound-1 -- Executing [EMAIL PROTECTED]:1] Directory(IAX2/vitel-inbound-1, default|cynic-main) in new stack == Parsing '/etc/asterisk/voicemail.conf': Found -- IAX2/vitel-inbound-1 Playing 'dir-intro' (language 'en') Timestamp: 10027ms SCall: 1 DCall: 00023 [64.2.142.31:4569] Rx-Frame Retry[ No] -- OSeqno: 008 ISeqno: 006 Type: DTMF_E Subclass: 8 Timestamp: 11683ms SCall: 00023 DCall: 1 [64.2.142.31:4569] Tx-Frame Retry[-01] -- OSeqno: 006 ISeqno: 009 Type: IAX Subclass: ACK Timestamp: 11683ms SCall: 1 DCall: 00023 [64.2.142.31:4569] Rx-Frame Retry[ No] -- OSeqno: 009 ISeqno: 006 Type: DTMF_E Subclass: 7 Timestamp: 11963ms SCall: 00023 DCall: 1 [64.2.142.31:4569] Tx-Frame Retry[-01] -- OSeqno: 006 ISeqno: 010 Type: IAX Subclass: ACK Timestamp: 11963ms SCall: 1 DCall: 00023 [64.2.142.31:4569] Rx-Frame Retry[ No] -- OSeqno: 010 ISeqno: 006 Type: DTMF_E Subclass: 2 Timestamp: 12263ms SCall: 00023 DCall: 1 [64.2.142.31:4569] Tx-Frame Retry[-01] -- OSeqno: 006 ISeqno: 011 Type: IAX Subclass: ACK Timestamp: 12263ms SCall: 1 DCall: 00023 [64.2.142.31:4569] -- IAX2/vitel-inbound-1 Playing 'dir-nomatch' (language 'en') #=== DTMF starts to fail somewhere after 8 and before 2. The playback was interrupted when I pressed 8, but 872 should match, so it didn't get all the way to 2. All further DTMF in the call was likewise ignored and I was unable to even interrupt playback with further key presses. #=== #=== # Complete call #=== #=== spaceheater*CLI iax2 set debug IAX2 Debugging Enabled Really destroying SIP dialog '[EMAIL PROTECTED]' Method: NOTIFY Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Really destroying SIP dialog '[EMAIL PROTECTED]' Method: OPTIONS Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: NEW Timestamp: 00013ms SCall: 00023 DCall: 0 [64.2.142.31:4569] VERSION : 2 CALLED NUMBER : XX CODEC_PREFS : (gsm|ulaw|speex|ilbc|alaw|g729) CALLING NUMBER : XX CALLING PRESNTN : 0 CALLING TYPEOFN : 0 CALLING TRANSIT : 0 CALLING NAME: XX LANGUAGE: en
[asterisk-users] Re: asterisk-users Digest, Vol 33, Issue 13
Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] adding chan_celliax
I was unaware of that build, thanks Tzafrir! But it seems old... You can download the current complete sources with: svn checkout http://www.celliax.org:8081/svn/celliax/branches/test1 test1 or you can download a livecd from www.celliax.org and test it without install anything. For compiling yourself you actually just need the files that build chan_celliax, that you find in test1/celliax_stuff/ (chan_celliax.c chan_celliax_spandsp.c chan_celliax_spandsp.h) and to modify the channels/Makefile And, of course, the configuration files (particularly celliax.conf) that you find in celliax_stuff/newconfigs/*. So, if you would better like to compile with the asterisk-dev packages, just download the svn sources as told, put those three files in the asterisk/channels directory and modify the asterisk/channels/Makefile. Anyway, if you download and build directly from the svn sources (just make install from the test1/asterisk-1.2.17 directory), it will put all the stuff in /usr/local/asterisk, /usr/local/asterisk/etc/asterisk/*.conf, /usr/local/asterisk/usr/sbin/*, etc, so it will not clutter your computer and other existing asterisk installations (you will have to remove just the /usr/local/asterisk directory). The test1 branch of the svn is the latest and greatest, but do not yet support skype and alsavoicemodems. The trunk of the svn supports skype and alsavoicemodems, but... ;-) Giovanni On 4/3/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Tue, Apr 03, 2007 at 08:50:02AM +0200, Giovanni Maruzzelli wrote: Ciao Patricio, have you compiled Asterisk from sources? At the moment you can only add chan_celliax support if you compiled from source. If this is the case, I can give you full instruction. And if from packages: http://updates.xorcom.com/contrib/celliax/ could be a good start. Though I only gotten it to build. Generally there shouldn't be a problem building chan_celliax vs. the package asterisk-dev . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] ZAP device reference in Zaptel 1.4
Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing [EMAIL PROTECTED]:1] SetCDRUserField(SIP/103-b7802230, Telstra) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/103-b7802230, ZAP/4/69223139) in new stack [Apr 4 10:47:43] WARNING[5659]: channel.c:3024 ast_request: No channel type registered for 'ZAP' [Apr 4 10:47:43] WARNING[5659]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) == Everyone is busy/congested at this time (1:0/0/1) == Auto fallthrough, channel 'SIP/103-b7802230' status is 'CHANUNAVAIL' Zaptel Version: 1.4.1 Echo Canceller: MG2 Configuration == Channel map: Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 2 channels configured. -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Adding DND to dialplan
Brian McEntire wrote:
Hello -
I've read Asterisk should be able to activate a do not disturb feature
Instead of using 2 extensions, you can get away with just one. Check
the database entry at the start, if it's already set, remove it. If
it's not there, add it.
[dnd]
; **
; Do not disturb can be set via Asterisk
; instead of the phones by dialing this
; number.
; **
exten = 79*,1,Set(CALLBACK=${DB(DND/${CALLERIDNUM})})
exten = 79*,2,GotoIf($[${CALLBACK} = YES]?79*,3:79*,101)
exten = 79*,3,Set(DB(DND/${CALLERIDNUM})=NO)
exten = 79*,4,Playback(local/stutter)
exten = 79*,5,Playback(de-activated)
exten = 79*,6,Hangup()
exten = 79*,101,Set(DB(DND/${CALLERIDNUM})=YES)
exten = 79*,102,Playback(local/stutter)
exten = 79*,103,Playback(activated)
exten = 79*,104,Hangup()
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve neither Liberty nor Safety.
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Re: [asterisk-users] ZAP device reference in Zaptel 1.4
Devraj Mukherjee wrote: Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing [EMAIL PROTECTED]:1] SetCDRUserField(SIP/103-b7802230, Telstra) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/103-b7802230, ZAP/4/69223139) in new stack [Apr 4 10:47:43] WARNING[5659]: channel.c:3024 ast_request: No channel type registered for 'ZAP' [Apr 4 10:47:43] WARNING[5659]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) You need to reinstall Asterisk. You installed Asterisk before installing Zaptel so Asterisk did not build anything that requires Zaptel. ___ --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] Interconnecting Cisco 1760 routers with Asterisk
Check this out HYPERLINK
javascript:ol('http://www.voip-info.org/wiki-Asterisk+cisco+FXO');http://w
ww.voip-info.org/wiki-Asterisk+cisco+FXO
_
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Joesph
Enviado el: Martes, 03 de Abril de 2007 02:53 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [asterisk-users] Interconnecting Cisco 1760 routers with Asterisk
Good day everyone.
I have Cisco 1760 routers that do site to site voip. Each router has 2 fxs
ports that connect to the local pbx and use sip to connect to other routers
over the WAN. I am thinking of putting in an asterisk box at the hub site
for interconnectivity with our global office voip provider. This provider
runs asterisk.
Question is - can Cisco 1760 routers make/receive calls to/fro asterisk? if
yes, any sample configuration please?
Thanks and regards
Joesph
Abuja, Nigeria
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[asterisk-users] Called Number Issue
Ok.. I have one box running Asterisk - Box1, and I'm trying to get another setup out on the internet (Box2) with an IAX2 trunk connecting the two. The calls flow fine from Box2 to Box1, but when I call Box2 from Box1 the Called Number always shows up as 's'. Why wont it pass the DID? Config in Box1: [ext-did] exten = 6222626,1,Set(FROM_DID=6222626) exten = 6222626,n,Goto(ext-local,6222626,1) [6222626] username=6222626 type=friend secret=6222626 qualify=no port=4569 host=dynamic context=from-internal callerid=User1 6222626 ___ --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] Faxing issues
I have spandsp, rxfax and asterisk-1.4.2 installed and whenever a fax call
comes in we get this. This isn't good. Any ideas?
[New Thread -1215390800 (LWP 8504)]
-- Accepting call from 'DELETED' to '539' on channel 0/1, span 1
-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1, DIALEDNUM=539) in
new stack
-- Executing [EMAIL PROTECTED]:2] Answer(Zap/1-1, ) in new stack
-- Executing [EMAIL PROTECTED]:3] Ringing(Zap/1-1, ) in new stack
-- Executing [EMAIL PROTECTED]:4] Wait(Zap/1-1, 4) in new stack
[New Thread -1215636560 (LWP 8505)]
[Thread -1215636560 (LWP 8505) exited]
-- Redirecting Zap/1-1 to fax extension
== Spawn extension (telco-incoming, fax, 0) exited non-zero on 'Zap/1-1'
-- Executing [EMAIL PROTECTED]:1] Goto(Zap/1-1,
internal-ext|fax-39|1) in new stack
-- Goto (internal-ext,fax-39,1)
-- Executing [EMAIL PROTECTED]:1] Macro(Zap/1-1,
faxmail|[EMAIL PROTECTED]) in new stack
-- Executing [EMAIL PROTECTED]:1] Set(Zap/1-1,
FAXFILE=/var/spool/asterisk/fax/1175648783.0.tif) in new stack
-- Executing [EMAIL PROTECTED]:2] Set(Zap/1-1, EMAILADDR=
[EMAIL PROTECTED]) in new stack
-- Executing [EMAIL PROTECTED]:3] RxFAX(Zap/1-1,
/var/spool/asterisk/fax/1175648783.0.tif) in new stack
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread -1215390800 (LWP 8504)]
0x08082599 in __ast_read (chan=0x9a17458, dropaudio=0) at channel.c:2128
2128if (AST_LIST_EMPTY(chan-readq) ||
!AST_LIST_NEXT(AST_LIST_FIRST(chan-readq), frame_list)) {
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Re: [asterisk-users] ZAP device reference in Zaptel 1.4
Hi Eric, Thanks for your suggestion I just reinstalled Asterisk, it still doesn't seem to know anything about Zaptel. I am using CentOS and installed Asterisk using yum from ATrpms. Anything else I can try? On 4/4/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Devraj Mukherjee wrote: Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing [EMAIL PROTECTED]:1] SetCDRUserField(SIP/103-b7802230, Telstra) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/103-b7802230, ZAP/4/69223139) in new stack [Apr 4 10:47:43] WARNING[5659]: channel.c:3024 ast_request: No channel type registered for 'ZAP' [Apr 4 10:47:43] WARNING[5659]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) You need to reinstall Asterisk. You installed Asterisk before installing Zaptel so Asterisk did not build anything that requires Zaptel. ___ --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 -- I never look back darling, it distracts from the now, Edna Mode (The Incredibles) ___ --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] Require only GSM Codec
Not every client supports gsm. Usually it's a good idea to put ulaw as well or you could get errors when neither side supports the same codec. disallow=all allow=gsm allow=ulaw -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (702) 979-2906 Fax: (212) 279-2906 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronaldo Zacarias Afonso Sent: Tuesday, April 03, 2007 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Require only GSM Codec Hi Sanjay, I'm not sure about that, but I think you can configure it in, for example, /etc/asterisk/sip.conf. There is an option that you configure for each channel like: only=gsm It instructs the sip protocol, that only gsm codec must be used. I hope it has helped you. Regards, Ronaldo. On 4/3/07, Sanjay Rajdev [EMAIL PROTECTED] wrote: Hello All, I would like to only use the gsm codec for all the calls, is it possible I want to use minimum possible bandwidth as we have most of calls over Internet. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ZAP device reference in Zaptel 1.4
From: Devraj Mukherjee [EMAIL PROTECTED] Date: Wed, 4 Apr 2007 11:46:11 +1000 Hi Eric, Thanks for your suggestion I just reinstalled Asterisk, it still doesn't seem to know anything about Zaptel. I am using CentOS and installed Asterisk using yum from ATrpms. Anything else I can try? Try lsmod to confirm that zaptel is indeed installed. I'm not familiar with CentOS or yum, but I assume you installed a binary package, so chan_zap.so is probably included. Hope this helps. Yuan Liu On 4/4/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Devraj Mukherjee wrote: Hi Everyone, I am using Zaptel and Asterisk 1.4 and have a Digium card with two FXS modules. The card works and ztcfg reports that it finds the two modules. Howevery when I try and place a call through the gateway I get the following error message. I have tried to refer to the ZAP device as ZAP/g2 etc Any suggestions? Anything that's different about Zaptel 1.4? -- Executing [EMAIL PROTECTED]:1] SetCDRUserField(SIP/103-b7802230, Telstra) in new stack -- Executing [EMAIL PROTECTED]:2] Dial(SIP/103-b7802230, ZAP/4/69223139) in new stack [Apr 4 10:47:43] WARNING[5659]: channel.c:3024 ast_request: No channel type registered for 'ZAP' [Apr 4 10:47:43] WARNING[5659]: app_dial.c:1090 dial_exec_full: Unable to create channel of type 'ZAP' (cause 66 - Channel not implemented) You need to reinstall Asterisk. You installed Asterisk before installing Zaptel so Asterisk did not build anything that requires Zaptel. ___ --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
