[Asterisk-Users] Supervised transfer over SIP to outside POTS lines
Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P-rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the global office. So the help desk support on the Suse machines need to transfer a call to an available local rep in another state. I thought this was possible until I realized the transfer only works on xPRO, which isn't available for linux. So I cant rely on SIP to handle this, I set up my extensions.conf have transfers, ie: [sip-exten] exten = 1001,1,Dial(SIP/1001,20,Trt) exten = 1001,2,Hangup And features.conf is : [featuremap] blindxfer = *1; Blind transfer ;disconnect = *0 ; Disconnect ;automon = *1 ; One Touch Record atxfer = *2 ; Attended transfer OK each analog phone has three way calling on it ... can I set up a flash command? How would that be done???. Thanks so much!! D ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Dev] RE: [Asterisk-Users] Zaptel update, Asterisk 1.2 janitor projects
On Tue, 2005-07-26 at 20:11 +0200, Michiel van Baak wrote: On 18:30, Tue 26 Jul 05, Dave Cotton wrote: On Tue, 2005-07-26 at 18:11 +0300, Tzafrir Cohen wrote: I suppose you refer to: http://lists.digium.com/pipermail/asterisk-cvs/2005-July/007125.html How do I track only the changes to the stable branch? For a user of Stable most of the messages on the CVS list are rather irrelevant. There seems to be a ' Tag: v1-0' in the message but it is in the body. Can't mutt filter on the body contents? it can. So the original problem can be solved :). -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] re: switch statement in dialplan
hi all, is there a switch statement in the dialplan? or do i have to daisy-chain GoToIf statements? i don't see a switch statement on the wiki, but you never know... thanks yair ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] super high bandwidth codec
On Tue, Jul 26, 2005 at 10:42:35PM -0700, Andrew C. Brown wrote: A recent blog entry indicated that GIPS was issuing licenses for its technology from a mere $50k for unlimited licenses with respect to an agreement with Microsoft. I don't have a huge concern about bandwidth limits. If I could get better quality than G.711 in the same bandwidhth that would be great. However, since I'm using IAX2 based DIDs and termination would it really matter? That is, if the ITSPs are connection to the PSTN via TDM interconnects wouldn't any calls be limited to G.711 quality anyway? IAX2 is a protocol, not a codec, so has little impact on sampling quality. But the second assumption is correct. If you are going to PSTN at any point in the chain, you are back to 8kHz sample rate and that extra spectrum you put over iSAC or whatever is tossed out the window. And also when you use MeetMe, right? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] qozap junghanns errors
Hi Altus, this seems the same error I got from my server and I'm interested to solve it but I have a TDM400P and a monoBRI junghanns compatible card. That error arise due to a interrupt confict I cannot resolve. How many cards have you got on your PC? TIA Giorgio Altus Snyman wrote: Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 18 z1 108 z2 74 As far as I know this is a motherboard error,I change the motherboard and it was working Its asterisk 1.0.9 and bristuff-0.2.0-RC8j Any Ideas please ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some more VOICEMAILMAIN issue...
On Tue, Jul 26, 2005 at 08:55:41AM +0200, Mauro Zanin wrote:
Hi everybody,
I have corrected this line in extensions.conf by stripping spaces off and now
it executes:
exten = 22999,1,VoiceMailMain(s${CALLERIDNUM})
And did you create voicemail-boxes for the relevant callers? In
voicemail.conf normally.
--
Tzafrir Cohen | [EMAIL PROTECTED] | VIM is
http://tzafrir.org.il | | a Mutt's
[EMAIL PROTECTED] | | best
ICQ# 16849755 | | friend
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[Asterisk-Users] echo capi AVM fritz card
Hi All, I'm running asterisk 1.0.7 on debian sarge, and hardware-wise an AVM Fritz!PCI Card, together with chan_capi 0.3.5. The problem is that any inbound/outbound calls on analogue line result in echo on MY end (the asterisk end). I've played with the echo settings in capi.conf (mainly turning on echocancel and echosquelch, also tried playing with rxgain/txgain) to no avail. The only setting that has helped (somewhat) so far is enabling echosquelch. The echo disappears but a new problem arises. When the person on the other end starts to talk, the first bit is chopped off, and the last bit (before they go quiet) so it almost sounds as though it's doing voice detection and transmitting only when it detects voice. Also, if the other person is talking and i start to talk, they get cut off immediately so this isn't a practical workaround. Any help will be muchly appreciated. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] qozap junghanns errors
Only one card:-) This is the second time I had it on a Intel board Even junghanns does not know about it Giorgio Incantalupo wrote: Hi Altus, this seems the same error I got from my server and I'm interested to solve it but I have a TDM400P and a monoBRI junghanns compatible card. That error arise due to a interrupt confict I cannot resolve. How many cards have you got on your PC? TIA Giorgio Altus Snyman wrote: Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 18 z1 108 z2 74 As far as I know this is a motherboard error,I change the motherboard and it was working Its asterisk 1.0.9 and bristuff-0.2.0-RC8j Any Ideas please ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mpg123 - two processes
I noticed this, but then I moved to madplay which only uses 1 process. Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber Sent: 27 July 2005 03:38 To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mpg123 - two processes Yes, I always have two. MARK. Billy Dunn wrote: Does everyone have two processes running for mpg123? I always have them when I'm running an idle Asterisk box. No calls going in or out and nothing off hook. Is this normal? Thanks! 5008 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-ri 5015 ?S 0:00 /usr/sbin/asterisk 5061 ?S 0:00 mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-ri ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 ### This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange. For more information, connect to http://www.f-secure.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_mysql does not write to mysql db
Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: US CallerID and TDM04B
Hi Rich, All,
Thanks for the response but this didn't help,
sorry.
I must be doing something wrong, it is very strange
that callerID is not working with one of largest US
Telco.
BTW, when I plug in my old Panasonic phone in the
same line, callerID works just fine.
Does anyone use SBC Telco (http://www.sbc.com/) and gets CallerID
successfully with asterisk? Could you drop me your
zapata.conf and zaptel.conf files?
Thanks in advance,
Boris Zolotarev
[EMAIL PROTECTED]
The only other item I can think of is to play with rxgain in
zapata.conf. Try rxgain=3.0, then rxgain=6.0, and if that seems to
impact receiving callerid, then adjust rxgain to the lowest value
where callerid still works.
Thanks for the response but still no
luck. I added those two lines just after the [channel] and updated my
dial plan but the result is the same (there is no CallerID):
Asterisk Ready. -- Starting simple switch on
'Zap/1-1' Jul 26 00:43:34 NOTICE[8867]: chan_zap.c:5367 ss_thread: Got
event 2 (Ring/Answered)... -- Executing
Wait("Zap/1-1", "2") in new stack -- Executing
NoOp("Zap/1-1", "") in new stack -- Executing
SetVar("Zap/1-1", "dnis=100") in new stack ... I
hope you have some more ideas, please? BTW, is there
anyone using Asterisk with SBC Telco in the US? Does your asterisk
recognize CallerID? Thanks, Boris
Zolotarev [EMAIL PROTECTED]
I have new TDM04B installed and
working fine with Asterisk 1.0.5 built on RedHat 9.
All is working fine except CallerID that bothers me big time. I
have several Panasonic and Sony phones and CallerID works fine with it (when I
plug in the line into phone instead into Asterisk I get
CallerID) but fails with Asterisk.I am based in
California (San Francisco) and my Telco is SBC (http://www.sbc.com/). . I would
really appreciate if anyone could take a look below at my zapata.conf
and extensions.conf and let me know what is wrong.
:: zapata.conf ::
[channels] Try adding these right here (right after [channels].
Let's see if that has any impact on the problem.
cidsignalling=bell
cidstart=ring
context=default switchtype=national
signalling=fxs_ks usecallerid=yes
callerid=asreceived hidecallerid=no
callwaiting=yes usecallingpres=yes
callwaitingcallerid=yes threewaycalling=yes
transfer=yes cancallforward=yes callreturn=yes
echocancel=yes echocancelwhenbridged=yes
echotraining=400 rxgain=0.0 txgain=0.0
group=1 callgroup=1 pickupgroup=1
immediate=no busydetect=yes busycount=7
musiconhold=default faxdetect=both context=zap
group=1 channel = 1-4
:: extensions.con ::I use this part of the code
to trace Asterisk log and check CallerID and CallerIDName.
[zap] exten = s,1,Wait(2) exten =
s,2,Answer() exten = s,3,SetVar(dnis=100) exten
= s,4,NoOp,${CALLERID} exten =
s,5,NoOp,${CALLERIDNAME} For incoming Zap calls (from the TDM
card), you do not need to "answer" the call unless you're going into an
IVR. If you are simply ringing a sip phone, do something like
this: [zap] exten =
s,1,NoOp,${CALLERID}
exten = s,2,Dial(Sip/1234,15)
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Re: [Asterisk-Users] qozap junghanns errors
Hi Altus, sorry about it. Have you tried to disable all you don't need on your server, for example parallel ports, serial ports, usb ports, etc?? Have you checked with cat /proc/interrups ?? Maybe your card share some interupt with other cards (eth0 for example). We are using Dell PCs but they do not let us to choose how to set interrupts, maybe your PC can. I'm sorry I cannot be more exaustive but this kind of problem is very hard to solve. Giorgio. Altus Snyman wrote: Only one card:-) This is the second time I had it on a Intel board Even junghanns does not know about it Giorgio Incantalupo wrote: Hi Altus, this seems the same error I got from my server and I'm interested to solve it but I have a TDM400P and a monoBRI junghanns compatible card. That error arise due to a interrupt confict I cannot resolve. How many cards have you got on your PC? TIA Giorgio Altus Snyman wrote: Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 18 z1 108 z2 74 As far as I know this is a motherboard error,I change the motherboard and it was working Its asterisk 1.0.9 and bristuff-0.2.0-RC8j Any Ideas please ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P Cable Pin Out
PRI CARD PINS 1 GREEN 2 RED 3 4 YELLOW 5 BLACK 6 7 8 PROVIDER PINS 1 2 3 YELLOW 4 GREEN 5 RED 6 BLACK 7 8 this setting work for us. We are located in slovenia, europe. On 7/27/05, Paul Dracevich [EMAIL PROTECTED] wrote: I have just got a TE110P card, and I need the cable pin out. Thanks -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 7/25/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql does not write to mysql db
Dear Kib, As I believe the Realtime options concerning the mysql database can only be used with the Asterisk CVS-HEADversion it's still not implemented on Asterisk v 1.0.* . Thx MAG Kib Eki wrote: Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] TE110P Cable Pin Out
I think you can use normal network cable, I'm from Italy and it's work perfectly Good luck -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Miloš Kocbek Inviato: mercoledì 27 luglio 2005 10.04 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] TE110P Cable Pin Out PRI CARD PINS 1 GREEN 2 RED 3 4 YELLOW 5 BLACK 6 7 8 PROVIDER PINS 1 2 3 YELLOW 4 GREEN 5 RED 6 BLACK 7 8 this setting work for us. We are located in slovenia, europe. On 7/27/05, Paul Dracevich [EMAIL PROTECTED] wrote: I have just got a TE110P card, and I need the cable pin out. Thanks -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 7/25/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Does Asterisk need to know where the call is comming from?
Hi: When a system like cisco is connected to the pstn through an asterisk server, does it matterfor Asterisk what system is making the request as long as the username, password, and supported protocol are correct? In other words, if the system is using the right protocol, sip or IAX, and the right authentication, is there anything else needed by asterisk so it can pass the call? Thanks. Start your day with Yahoo! - make it your home page ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX over HTTP
Rob == Rob Scott [EMAIL PROTECTED] writes: Rob For work environments where you only get HTTP or HTTPS access, Rob what is the feasibility of doing IAX over HTTP? Tunnelling tcp/ip over http/(tls/)?tcp/ip is viable, tunnelling rtp/udb/ip or iax/udp/ip over http/(tls/)?tcp/ip however will only work reliably if the tcp doesn't see any packet loss. Else it will retransmit lost packets and the voice quality will suck. That said, if you can get a http or https socket going you can probably also tunnel over dns. So you may want to look into ip over dns/udp/ip tunnels for rtp or iax. -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] I found problem with TE110P and the new kernel of fedora, kernel panic
I installed a new fedora 3 and i did the yum update, In this way I upgrade the kernel 2.6.11-35, before I used the 2.6.9-77 for fedora and it was work perfectly no problem. When I made the upgrade I got the kernel panic every time that I remove the drivers or restart the computer. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] qozap junghanns errors
Giorgio Incantalupo wrote: Thanks Will have a look Hi Altus, sorry about it. Have you tried to disable all you don't need on your server, for example parallel ports, serial ports, usb ports, etc?? Have you checked with cat /proc/interrups ?? Maybe your card share some interupt with other cards (eth0 for example). We are using Dell PCs but they do not let us to choose how to set interrupts, maybe your PC can. I'm sorry I cannot be more exaustive but this kind of problem is very hard to solve. Giorgio. Altus Snyman wrote: Only one card:-) This is the second time I had it on a Intel board Even junghanns does not know about it Giorgio Incantalupo wrote: Hi Altus, this seems the same error I got from my server and I'm interested to solve it but I have a TDM400P and a monoBRI junghanns compatible card. That error arise due to a interrupt confict I cannot resolve. How many cards have you got on your PC? TIA Giorgio Altus Snyman wrote: Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 18 z1 108 z2 74 As far as I know this is a motherboard error,I change the motherboard and it was working Its asterisk 1.0.9 and bristuff-0.2.0-RC8j Any Ideas please ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Monitoring
Can anyone help me how to open recorded converstations in asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql does not write to mysql db
Kib Eki a crit: Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Asterisk don't use directly mysql database for cdr, astersisk use odbc and odbc connect to mysql. So you must configure odbc corectly wiyt libmyodbc (on debian) the config file are here: /etc/odbcinst.ini : [MySQL] Description = MySQL driver Driver = /usr/lib/odbc/libmyodbc.so Setup = /usr/lib/odbc/libodbcmyS.so CPTimeout = CPReuse = FileUsage = 1 /etc/odbc.ini : [MySQL-asterisk] Description = MySQL Asterisk Database Driver = MySQL Socket = /var/run/mysqld/mysqld.sock Server = @ipofMysqlddatabase (not domain name) User = Password = Database = asterisk Option = 3 #Port = /etc/asterisk/cdr_odbc.conf : [global] dsn=MySQL-asterisk username=database_username password=database_password loguniqueid=yes ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regarding Call Hold
Please file a bug report with a full SIP DEBUG output file. Set debug to 4, verbosity to 4 and turn on SIP debugging. Upload that file as an attachment to the bug report and place the bug report in the SIP category. Thanks! /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6
hi All I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb ram, with g729 for i686 , (fedora 1). my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen otherparty realtime voice , but other party geting sip party's voice 1 sec later (not realtime). please some help me to solve this issu, last one month i am tring different different way to solve this issu. is it codec problem or something else. thanks bashir - Original Message - From: Aarthy G - CTD, Chennai. [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, July 27, 2005 1:12 AM Subject: [Asterisk-Users] Regarding Call Hold Hi All, We are using asterisk for testing our home gateway setup. We have implemented Call Hold feature in our application. In our Application we have written code in such a way that for an INVITE for putting a SIP phone on HOLD will contain connection address 0.0.0.0 in the SDP message. We expect the same connection address i.e 0.0.0.0 in the 200 OK response for the INVITE that is sent. This feature works when we tested without involving Asterisk. Now after configuring Asterisk as our Registrar and OutBound Proxy, we find that Call hold is not getting through. But we are getting a 200 0K with connection address as the host ip of Asterisk. We see that the this ReInvite is not getting forwarded to the appropriate detsination from the asterisk. We are not looking for music on hold feature. Output of sip debug and the two configuration files sip.conf and extensions.conf have been attached in this mail. Lines where we send 0.0.0.0 in the connection address field of SDP message and the 200 OK Response in which we get host ip of Asterisk in connection Address have been highlighted in RED in the attached word document. Please go through the configuration files and the debug output and suggest us the necessary changes that have to be done by us. We also do not want music_on_hold feature. Can somebody here please tell us about how to configure asterisk to disable music on hold and get 0.0.0.0 in the 200 OK response for the Re-Invite Sent? thanks, Aarthy G. Call-Hold.zip DISCLAIMER This message and any attachment(s) contained here are information that is confidential, proprietary to HCL Technologies and its customers. Contents may be privileged or otherwise protected by law. The information is solely intended for the individual or the entity it is addressed to. If you are not the intended recipient of this message, you are not authorized to read, forward, print, retain, copy or disseminate this message or any part of it. If you have received this e-mail in error, please notify the sender immediately by return e-mail and delete it from your computer. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P Cable Pin Out
1 - 4 2 - 5 4 - 1 5 - 2 http://www.voip-info.org/tiki-index.php?page=crossover+T1+cable - Original Message - From: Milos Kocbek [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 27, 2005 1:03 AM Subject: Re: [Asterisk-Users] TE110P Cable Pin Out PRI CARD PINS 1 GREEN 2 RED 3 4 YELLOW 5 BLACK 6 7 8 PROVIDER PINS 1 2 3 YELLOW 4 GREEN 5 RED 6 BLACK 7 8 this setting work for us. We are located in slovenia, europe. On 7/27/05, Paul Dracevich [EMAIL PROTECTED] wrote: I have just got a TE110P card, and I need the cable pin out. Thanks -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 7/25/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P Cable Pin Out
Paul Dracevich wrote: I have just got a TE110P card, and I need the cable pin out. http://www.gcom.com/home/support/t1crossover.html Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE110P Cable Pin Out
On Wed, 27 Jul 2005, Paul Dracevich wrote: I have just got a TE110P card, and I need the cable pin out. The TE110P cards use the standard T1/E1 modular pinout. See http://www.samhassan.com/isdn60.gif. 1 Receive from pstn (tip2) 2 Receive from pstn (ring2) 4 Transmit to pstn (ring1) 5 Transmit to pstn (tip1) Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are busy and congestion behaving differently than documented?
Steve Gladden wrote: exten = ,1,Answer exten = ,2,busy(35) exten = ,3,Hangup Asterisk plays a 'busy' signal for 35 seconds I have also tried this with congestion (instead of busy). What is strange is that in either case, the busy tones are coming from asterisk and *not* being locally generated by the PAP2-NA. If the channel is answered Asterisk has to do inband (audio) tones. --Eric -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why is sip saying NO NAT
In the sip conversation below, the traffic is sent to the client and Asterisk is saying noNAT However, this client is configured nat=yes qualify=yes canreinvite=no So why does it still say noNAT? The IP of the phone is 10.0.0.xx Retransmitting #4 (no NAT) to 209.59.xx.xx:10979: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 207.44.xx.xx::5060;branch=z9hG4bK3071ac05 From: asterisk sip:[EMAIL PROTECTED];tag=as121afa0c To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 27 Jul 2005 08:38:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN ASTERISK Cabling...
Hiya! I have a question about how to cable my asterisk together in Germany. Here is the setup: 1) Anlage anschluss (pbx line - 4 isdn lines bundled together) 2) These are delivered via 4 NTBAs 3) In my computer, I have asterisk and an AVM C4 card 4) I need to connect 8 isdn phones to connect the NTBAs to the computer - no problem. However, what would I need in order to connect asterisk to ISDN phones within the office? I need not ask that we are operating on a shoestring... thanks! Álainn The cheese-mites asked how the cheese got there, And warmly debated the matter; The orthodox said that it came from the air, And the heretics said from the platter. Anon. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel error: Unable to create channel op type 'Zap'
Hi, In struggeling with this problem for a two weeks now. I have a X100P clone card in my * box but I'm not able to get it to run. I'm running fedora core 3 with kernel 2.6.12-1.1372_FC3 on a VIA EPIAML500EA The compiling of both zaptel and asterisk went without any errors. I can run zaptel and asterisk without any errors. When I run ztcfg I don't get any errors too. But when I try to place a call trough my x100p I get this error message in asterisk: NOTICE app_dial.c:1091 dial_exec_full: Unable to create channel op type 'Zap' Outside calls are not comming in either. Here are my zapata.conf and zaptel.conf: -zapata.conf- [channels] signalling=fxs_ks context=incoming channel=1 -zaptel.conf- loadzone = nl defaultzone=nl fxsks=1 --- The funny part comes here: I'm installing a *box for a friend with a ISDN card and the same problem occures. So I probarbly doing something wrong in fedora... Any ideas??? Thanks, Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] I found problem with TE110P and the new kernel offedora, kernel panic
I had a problem with this card and 2.6.11 kernel. I am using FC3 but sticking with the 2.6.9 kernel. I had a lot of make warnings on the zaptel build and the card played up. It also wouldn't do a modprobe -r without crashing the system. With 2.6.9 zaptel compiles fine and I can unload the mod as and when. Also stay well away from the 2.6.12 FC3 kernel as it didn't work at all and didn't come with any sources. Regards Lee From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Yousef HerzallahSent: 27 July 2005 09:34To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] I found problem with TE110P and the new kernel offedora, "kernel panic" I installed a new fedora 3 and i did the yum update, In this way I upgrade the kernel 2.6.11-35, before I used the 2.6.9-77 for fedora and it was work perfectly no problem. When I made the upgrade I got the kernel panic every time that I remove the drivers or restart the computer. --No virus found in this incoming message.Checked by AVG Anti-Virus.Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 ###This message has been scanned by F-Secure Anti-Virus for Microsoft Exchange.For more information, connect to http://www.f-secure.com/ -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: sip+oh323 - no voice at sip side
On 26-07-2005 at 07:23:39PM +0200, [EMAIL PROTECTED] wrote: Hello, I have something like this: SIPUSER -sip- ASTERISK -oh323- AUDIOCODEC -e1- PSTN If I call from SIP to PSTN, at the beginning of the call (1 second) after getting phone at the PSTN side I hear voice at the SIP side. After this 1 second I don't hear anything in SIP phone (at the PSTN phone everything is OK). Nobody has had any problems like me? Bartek. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql does not write to mysql db
Try to put the IP of you CDR server instead of 'localhost', that's work for me. Regards. - Original Message - From: Kib Eki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 27, 2005 9:44 AM Subject: [Asterisk-Users] cdr_mysql does not write to mysql db Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: super high bandwidth codec
Eric Wieling aka ManxPower [EMAIL PROTECTED] uttered the following thing: Dean Collins wrote: I've just gotten off a skype conference call and it pisses me off that the quality of skype is higher than my asterisk calls. Is there such a thing as a super high bandwidth codec? Asterisk does not support wideband codecs as far as I know. Most It seems to - I have successfully used Speex 16KHz with Asterisk, even using voice prompts in this mode. Sounds a lot better than the same audio in a-law. With a softphone of course, in this case Eyebeam. Of course, the rules about PSTN breakout still apply... BB ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voice mailbox on the fly?
Hi All... I'm trying to figure out how to get Asterisk to answer a number, prompt the caller for a code 6 digit code and then prompt the caller to leave a message. I then want to email that message out. I realize this is not likelt t be readily available, but could someone offer a suggestion about how I might implement this? Could I do it with the existing Asterisk apps or do I have to write a new one? Thanks... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] super high bandwidth codec
Tzafrir Cohen wrote: On Tue, Jul 26, 2005 at 10:42:35PM -0700, Andrew C. Brown wrote: A recent blog entry indicated that GIPS was issuing licenses for its technology from a mere $50k for unlimited licenses with respect to an agreement with Microsoft. I don't have a huge concern about bandwidth limits. If I could get better quality than G.711 in the same bandwidhth that would be great. However, since I'm using IAX2 based DIDs and termination would it really matter? That is, if the ITSPs are connection to the PSTN via TDM interconnects wouldn't any calls be limited to G.711 quality anyway? IAX2 is a protocol, not a codec, so has little impact on sampling quality. But the second assumption is correct. If you are going to PSTN at any point in the chain, you are back to 8kHz sample rate and that extra spectrum you put over iSAC or whatever is tossed out the window. And also when you use MeetMe, right? I'm researching that. All I've been able to find so far is http://lists.digium.com/pipermail/asterisk-users/2005-May/107214.html which says that basically, no, Asterisk can't yet handle anything but 8KHz sample rates (though I suppose that doesn't necessarily preclude reinvited peer to peer VoIP calls where Asterisk removes itself from the audio path). If you find any more references on that issue, please post them. This question of high quality voice is going to keep coming up so I'd like there to be Wiki page to bring people up to date on all this we're discussing. And frankly I'd like to help build some momentum towards increased spectrum voice telephony. Right now, few people even think to ask and VoIP to them is just about saving money rather than improving the product. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WARNING[3240]: chan_oss.c:305 sound_thread: Read error on sound device: File descriptor in bad state
Hello everyone, I keep getting the warning message above all the time. Any clues on how to solve this problem? Thanks in advance, Leo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fritz PCI card in ptp mode with chan_misdn
Craig, You obviously have has experience with chan_mISDN in AU and the Fritz. Have you tried chan_capi? I am currently using a Fritz with chan_capi in AU and am not entirely happy with it. Is chan_mISDN any better? On 7/27/05, Craig Guy [EMAIL PROTECTED] wrote: The mISDN Fritz! driver supports PTP mode. In your startup script where you load the mISDN drivers call the fritz driver thusly: modprobe avmfritz protocol=34 Bit 5 sets PTP mode, bits 3-0 set the D-channel protocol ID (set bit one for DSS1). Craig - Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, July 27, 2005 12:35 AM Subject: Re: [Asterisk-Users] Fritz PCI card in ptp mode with chan_misdn On 18:21, Mon 25 Jul 05, Johann Steinwendtner wrote: Hello ! I would like to get working a Fritz PCI card using chan_misdn operating in ptp mode. As far as I know the fritz cards do not support ptp mode. We tried all the possible config file options with chan_capi and in the end we trashed them and installed a junghanns QuadBRI. If you get it working in ptp mode, please tell me how you did it. -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice mailbox on the fly?
Why not exten = 123,1,BackGround(whatIsthe6Digets) exten = 123456,1,Voicemail(u123456) Jim Archer wrote: Hi All... I'm trying to figure out how to get Asterisk to answer a number, prompt the caller for a code 6 digit code and then prompt the caller to leave a message. I then want to email that message out. I realize this is not likelt t be readily available, but could someone offer a suggestion about how I might implement this? Could I do it with the existing Asterisk apps or do I have to write a new one? Thanks... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel error: Unable to create channel op type 'Zap'
I just did the modprobe 2 times and it worked but that was on the 2.6.9 kernel Something about core 3 taking its time to create the device modprobe zaptel sleep 3 modprobe zaptel :-) Peter Raaijmakers wrote: Hi, In struggeling with this problem for a two weeks now. I have a X100P clone card in my * box but I'm not able to get it to run. I'm running fedora core 3 with kernel 2.6.12-1.1372_FC3 on a VIA EPIAML500EA The compiling of both zaptel and asterisk went without any errors. I can run zaptel and asterisk without any errors. When I run ztcfg I don't get any errors too. But when I try to place a call trough my x100p I get this error message in asterisk: NOTICE app_dial.c:1091 dial_exec_full: Unable to create channel op type 'Zap' Outside calls are not comming in either. Here are my zapata.conf and zaptel.conf: -zapata.conf- [channels] signalling=fxs_ks context=incoming channel=1 -zaptel.conf- loadzone = nl defaultzone=nl fxsks=1 --- The funny part comes here: I'm installing a *box for a friend with a ISDN card and the same problem occures. So I probarbly doing something wrong in fedora... Any ideas??? Thanks, Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call Monitoring
Hi, if the file format is a problem, try Wavepad, it could help you. Giorgio Ian Bert Tusil wrote: Can anyone help me how to open recorded converstations in asterisk? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice mailbox on the fly?
Thanks for the reply... Well I need the voice mail WAV file mailed to a different email address, depending upon what the code is. But this looks interesting: http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail The only problem I see with this is that the mailbox ID is only int(5) and I need 6 or 7... Maybe I could modify the voicemail app to read the data directly from my own database structure Thinking... --On Wednesday, July 27, 2005 2:18 PM +0200 Altus Snyman [EMAIL PROTECTED] wrote: Why not exten = 123,1,BackGround(whatIsthe6Digets) exten = 123456,1,Voicemail(u123456) Jim Archer wrote: Hi All... I'm trying to figure out how to get Asterisk to answer a number, prompt the caller for a code 6 digit code and then prompt the caller to leave a message. I then want to email that message out. I realize this is not likelt t be readily available, but could someone offer a suggestion about how I might implement this? Could I do it with the existing Asterisk apps or do I have to write a new one? Thanks... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent penalties and busy status
Hi all, When implementing a queue using members like this: [queue] strategy = rrmemory member = Agent/1000,1 member = Agent/1001,1 member = Agent/1002,2 member = Agent/1003,2 member = Agent/1004,2 And you call into the queue, agents 1000 and 1001 will ring in an alternating fashion until one of them answers it. You might have seen my question coming already, so I won't delay it anymore: is it possible to have 1000 and 1001 only ring once and then fallback to the other penalty-levels? With disciplined agents it's no problem, but when 1000 and 1001 decide to not answer any calls for a while (circuit-busy / noanswer), the rrmemory strategy doesn't fail over to other agents and the whole queue is stuck. If it isn't possible, I would be happy to change this in the Asterisk code for my site installation, but where should I start hacking? Any pointers are greatly appreciated. Thanks in advance for your time. With kind regards, Frank Schoep ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sendDTMF at pickup
Hi everyone:
I want to send a DTMFtone c after the person picks up the telephone, and
only if the person has pick up the phone.
The following configuration is sending the c tone after our prefix and the
beep, but before it dials the destiny.
[voip]
exten=1001,1,Dial(SIP/1001,60,tr)
exten=1002,1,Dial(SIP/1002,60,tr)
exten=502,1,Dial(SIP/502,60,tr)
exten=504,1,Dial(SIP/504,60,tr)
exten=i,1,NoCDR()
exten=i,2,Hangup()
exten=s,1,Wait(2)
exten=s,2,Background(beep||)
exten=s,3,DigitTimeout(6)
exten=s,4,ResponseTimeout(10)
exten=s,5,SendDTMF(c)
exten=t,1,NoCDR()
exten=t,2,Hangup()
exten=_009[13456789].,1,Dial(SIP/operador/${EXTEN},60,tr)
exten=_009[2].,1,Dial(SIP/operador/${EXTEN},60,tr)
exten=_00[12345678].,1,Dial(SIP/operador/${EXTEN},60,tr)
exten=_6[0123456789].,1,Dial(SIP/operador/${EXTEN},60,tr)
exten=_9[123456789].,1,Dial(SIP/operador/${EXTEN},60,tr)
tengo este resultado:
-- Executing Goto(Zap/2-1, discriminador|93185|1) in new stack
-- Goto (discriminador,93185,1)
-- Executing Goto(Zap/2-1, voip|s|1) in new stack
-- Goto (voip,s,1)
-- Executing Wait(Zap/2-1, 2) in new stack
-- Executing BackGround(Zap/2-1, beep||) in new stack
-- Playing 'beep' (language '')
-- Executing DigitTimeout(Zap/2-1, 6) in new stack
-- Set Digit Timeout to 6
-- Executing ResponseTimeout(Zap/2-1, 10) in new stack
-- Set Response Timeout to 10
-- Executing SendDTMF(Zap/2-1, c) in new stack
== CDR updated on Zap/2-1
-- Executing Dial(Zap/2-1, SIP/operador/66983|60|tr) in new stack
-- Called operador/66983
Note: Everything else works fine.
thanks
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R: [Asterisk-Users] Sound Quality Problems
Try to use 2.6.10 kernel not 2.6.12.3 kernel Coz I got some problems with the kernel 2.6.12, I had the same Digium Wildcard TE110P, Good luck Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Robert Christian Inviato: martedì 26 luglio 2005 19.13 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Sound Quality Problems Thanks for reading this. Ive been pulling hair for days trying to resolve this, and any help someone can give me would be very much appreciated. I have an Asterisk box that is basically a P4-3GHz, a Digium-recommended SuperMicro X5SSE-GM motherboard, 2GB RAM, 250GB IDE hard-drive with UDMA, a SoundBlaster Live! 24 sound card, a Digium Wildcard TE110P, and a Digium Wildcard TDM400P with 4 FXS modules. Right now the Ethernet is hooked up directly to a single VoIP phone (Grandstream GXP-2000). Im running kernel 2.6.12.3. THE PROBLEM: The problem I am having is when using the speakers hooked into the SB Live! 24 card. When I call through like an intercom (my intention) and have the dialplan play an announcement (for testing purposes, I have used both the included you-sound-cute and lots-o-monkeys), its very choppy to the point of being laughably unacceptable. Then when I talk over the intercom, my voice sounds just as choppy. Interestingly enough, when I go to the console and dial the demo included in the Asterisk sample configuration files (which I left in for testing), the womans voice sounds fine (for 8KHz anyway). But the demo is an IVR menu, and when I dial 2 for more information (for example), it starts playing the next message all choppy. I have read hundreds of mailing list posts and dozens of how-tos and diagnostics suggestions online. Ive tweaked out motherboard settings, kernel options, hard drive parameters, etc. with no success. Ive tried literally dozens of ALSA configurations thinking that the problem could be there. Ive focused on kernel and IRQ optimizing, ALSA configuration, Asterisk configuration, and Zaptel configuration (in case its a timing problem) with no success. I have been completely unable to resolve this problem. I have a PRI line on its way, but since zttest suggests the internal timing of the TE110P is accurate within Digium specs, I dont think external synchronization will help anything. Maybe the SB Live! 24 isnt the best card to use for the console intercom on this system. But supposedly ALSA has full support for it. Any help at all is greatly appreciated. Thank you. - Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap'
I tend to make it pause for 10 secs when loading the module as I have had a few occurances of loading before /dev/zap has been populated. Wouldn't trust the 2.6.12 kernel as far as I could virutally throw it. Has anyone had any problems with PCI-X systems? In particular call dropping? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Altus Snyman Sent: 27 July 2005 12:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap' I just did the modprobe 2 times and it worked but that was on the 2.6.9 kernel Something about core 3 taking its time to create the device modprobe zaptel sleep 3 modprobe zaptel :-) Peter Raaijmakers wrote: Hi, In struggeling with this problem for a two weeks now. I have a X100P clone card in my * box but I'm not able to get it to run. I'm running fedora core 3 with kernel 2.6.12-1.1372_FC3 on a VIA EPIAML500EA The compiling of both zaptel and asterisk went without any errors. I can run zaptel and asterisk without any errors. When I run ztcfg I don't get any errors too. But when I try to place a call trough my x100p I get this error message in asterisk: NOTICE app_dial.c:1091 dial_exec_full: Unable to create channel op type 'Zap' Outside calls are not comming in either. Here are my zapata.conf and zaptel.conf: -zapata.conf- [channels] signalling=fxs_ks context=incoming channel=1 -zaptel.conf- loadzone = nl defaultzone=nl fxsks=1 --- The funny part comes here: I'm installing a *box for a friend with a ISDN card and the same problem occures. So I probarbly doing something wrong in fedora... Any ideas??? Thanks, Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax detection on isdn
Hi all, I have previously got fax working on a digium card, now im using a sirrix isdn card. The problem is that it cant detect when it is a fax. So how do i get fax detection working on this sirrix isdn card. thanks, yusuf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call rejected/No application dial
I am getting forbidden Call rejected 403 errors in SJ Phone and on CLI I am getting NO application Dial for extension 102. 18:17:06 INFO Session rejected. Reason: 403 Forbidden 18:17:06 INFO Call 39 ended: Forbidden Call rejected: 403 Forbidden 18:17:06 INFO SIP: Session terminated. 18:17:13 DEBUG 2005-07-27 13:17:13.801 UDP LOCAL-10.1.6.18:5060 OPTIONS sip:10.1.6.18:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.6.22;rport;branch=z9hG4bK0a010616001042e7895935ce0044 Content-Length: 0 Call-ID: [EMAIL PROTECTED] CSeq: 17 OPTIONS From: sip:[EMAIL PROTECTED]:5060;tag=20060921811401 Max-Forwards: 70 To: sip:10.1.6.18:5060 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap'
I got the same problem yesterday and i just ch'ange the kernel from 2.6.12-1.1372_FC3 to 2.6.9 kernel of fedora core 3, and now it work perfectly. I think there is some problems with the new kernel of fedora. -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Altus Snyman Inviato: mercoledì 27 luglio 2005 13.45 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap' I just did the modprobe 2 times and it worked but that was on the 2.6.9 kernel Something about core 3 taking its time to create the device modprobe zaptel sleep 3 modprobe zaptel :-) Peter Raaijmakers wrote: Hi, In struggeling with this problem for a two weeks now. I have a X100P clone card in my * box but I'm not able to get it to run. I'm running fedora core 3 with kernel 2.6.12-1.1372_FC3 on a VIA EPIAML500EA The compiling of both zaptel and asterisk went without any errors. I can run zaptel and asterisk without any errors. When I run ztcfg I don't get any errors too. But when I try to place a call trough my x100p I get this error message in asterisk: NOTICE app_dial.c:1091 dial_exec_full: Unable to create channel op type 'Zap' Outside calls are not comming in either. Here are my zapata.conf and zaptel.conf: -zapata.conf- [channels] signalling=fxs_ks context=incoming channel=1 -zaptel.conf- loadzone = nl defaultzone=nl fxsks=1 --- The funny part comes here: I'm installing a *box for a friend with a ISDN card and the same problem occures. So I probarbly doing something wrong in fedora... Any ideas??? Thanks, Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice mailbox on the fly?
Jim Archer wrote: I'm trying to figure out how to get Asterisk to answer a number, prompt the caller for a code 6 digit code and then prompt the caller to leave a message. I then want to email that message out. I realize this is not likelt t be readily available, but could someone offer a suggestion about how I might implement this? Could I do it with the existing Asterisk apps or do I have to write a new one? Why not use the following: Authenticate Record System See the documentation for each application. You would use System to send the sound file using whatever method you want. I use mutt. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql does not write to mysql db
sylvain garcia wrote: Kib Eki a écrit : Asterisk don't use directly mysql database for cdr, astersisk use odbc and odbc connect to mysql. So you must configure odbc corectly wiyt libmyodbc (on debian) the config file are here: Wrong. Asterisk can and does connect to MySQL directly. (Where the hell are these people getting this wrong info?) -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Port Restricted CONE NAT Error
Dear All, I am getting Port Restricted CONE NAT Error in SJ Phone. I am using stun.softjoys.com as STUN value. Should I change it to my local server address or IP? Plus when I make a call I get a message Forbidden Call Rejected 403 Any ideas! Afzaal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cannot find channel_pvt.h
wassim darwish wrote: when i tried to compile asterisk-oh323 i get an error that channel_pvt.h is missing,where i can find and download it and in which directory i must put it. channel_pvt.h has been deprecated for quite some time. You need to update your 323 source as it shouldn't be using that header. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Melting TDM card
I had the problem on a very old version of the TDM card (brown card) I contacted Digium and after a few WTF's they sent me a shiny new blue card that to this day is still blue!!! Contact your reseller or Digium directly they always stand behind their products. I would double check for ground loops and good connectivity on the molex connector. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert ChristianSent: Tuesday, July 26, 2005 9:28 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Melting TDM card Yesterday FedEx brought me my new TDM400P with 4 FXS modules. I installed it (in the correct type PCI slot) , plugged in the power, and fired up the system. A few minutes later everyone in the office is complaining about something burning. I open the server again, and the top of the Digium card is black, slightly deformed, and looks and smells of melted plastic. The funny thing isthe card still works just fine (as far as my testing has revealed, anyway. I havent tried ringing the lines, just placing calls from them). Maybe this is how Digium knows if a card is returned used or not? Still, nothing was touching the card, it was in the right slot, and it was installed as instructed. Beats me what happened. Im just not sure I like spending hundreds of dollars on a card that arcs and melts when I install it even if it continues to work. Has anyone else had this problem with the TDM cards? - Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Real-time for H.323?
Ronald_Wiplinger wrote: Matthew, can we use real-time also for H.323 phones? (h323_buddies) ??? bye Ronald I don't see any realtime code in either H323 source. So, No. Not until someone adds it. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Klicking sounds in background
Hello, I have an IVR with Intel HMP SIP stack, which is a peer behind my Asterisk box (Asterisk 1.0.7, Digium PRI). When dialing in via PSTN, there are klicking sounds in the background, which do not appear, when dialing in via SIP (using Asterisk as pbx). The issue does not seem to be an alaw/µlaw problem. I tried trunking two Asterisk boxes via IAX and then call via two asterisks, but the same effect appears. Whenever there is PSTN involved, I have these klicking sounds, when there is no PSTN, everything works correctly. The setup works great with different SIP peers (others than the Intel...) Anyone has an idea? Best regards Jochen -- Jochen Witte email: [EMAIL PROTECTED] web: http://alpha-lab.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Attended transfer not working (atxfer)
While on conversation with another party, I dial the atxfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep' (language 'en') And then in the console of asterisk is wrote : -- Executing Hangup(Transfered/SIP/8008-432aZOMBIE, ) in new stack Number SIP/8008 is the first originator of the call which must be connected to the transferee. Any ideas? I use CVS of asterisk from 2005-06-16 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap'
I worked over two days setting up a card on fc 3, I had 2.6.9 installed. No action on the TE110P at all. I downloaded the stable zaptel/asterisk CVS and found an uncompiled wx110xp driver and recompiled. It starts up great now. There are some issues with fc3 FC3 has some issues with loading On Wed, 2005-07-27 at 15:01 +0200, Yousef Herzallah wrote: I got the same problem yesterday and i just ch'ange the kernel from 2.6.12-1.1372_FC3 to 2.6.9 kernel of fedora core 3, and now it work perfectly. I think there is some problems with the new kernel of fedora. -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Altus Snyman Inviato: mercoledì 27 luglio 2005 13.45 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap' I just did the modprobe 2 times and it worked but that was on the 2.6.9 kernel Something about core 3 taking its time to create the device modprobe zaptel sleep 3 modprobe zaptel :-) Peter Raaijmakers wrote: Hi, In struggeling with this problem for a two weeks now. I have a X100P clone card in my * box but I'm not able to get it to run. I'm running fedora core 3 with kernel 2.6.12-1.1372_FC3 on a VIA EPIAML500EA The compiling of both zaptel and asterisk went without any errors. I can run zaptel and asterisk without any errors. When I run ztcfg I don't get any errors too. But when I try to place a call trough my x100p I get this error message in asterisk: NOTICE app_dial.c:1091 dial_exec_full: Unable to create channel op type 'Zap' Outside calls are not comming in either. Here are my zapata.conf and zaptel.conf: -zapata.conf- [channels] signalling=fxs_ks context=incoming channel=1 -zaptel.conf- loadzone = nl defaultzone=nl fxsks=1 --- The funny part comes here: I'm installing a *box for a friend with a ISDN card and the same problem occures. So I probarbly doing something wrong in fedora... Any ideas??? Thanks, Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some more VOICEMAILMAIN issue...
U are using SIP ? if yes set *type=friend*
Bruno.
Mauro Zanin wrote:
Hi everybody,
I have corrected this line in extensions.conf by stripping spaces off
and now it executes:
*exten = 22999,1,VoiceMailMain(s${CALLERIDNUM})*
when it runs, the mail box number is asked and password too. I
expected no question were made, because I inserted CALLERIDNUMBER and
s in front of box number.
Anybody knows why?
Thank to you all, very kind members of this list!
Ciao
Mauro
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Re: [Asterisk-Users] cdr_mysql does not write to mysql db
using localhost in you mysql conf should work, make sure you linux box and the loopback interface up and has a a entry in your /etc/ hosts for localhost and that your firewall (if you have one setup on your localbox) allows traffic from 172.0.0.1 to 172.0.0.1 On Jul 27, 2005, at 6:17 AM, Dpto. Técnico. wrote: Try to put the IP of you CDR server instead of 'localhost', that's work for me. Regards. - Original Message - From: Kib Eki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 27, 2005 9:44 AM Subject: [Asterisk-Users] cdr_mysql does not write to mysql db Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql does not write to mysql db
Mohamed A. Gombolaty wrote: Dear Kib, As I believe the Realtime options concerning the mysql database can only be used with the Asterisk CVS-HEAD version it's still not implemented on Asterisk v 1.0.* . Thx MAG Wrong. He's not using RealTime. No where in his original post did he mention realtime. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 Configuration file
Folks! I would appreciate if someone could send me a simple working h323 configuration file oh323.conf that is part of [EMAIL PROTECTED] installation. I have tried to use the oh323.conf content listed on WIKI but it is just not working as my H323 endpoint ( PA168 based ATCOM Phone) cannot register. I need a working example of this file for similar phone. Seshu NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql does not write to mysql db
Kib Eki wrote: Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? Do you have the module installed? Is the module loaded? What happens when you type cdr mysql status on asterisk command line? You should see something like this: cytrex*CLI cdr mysql status Connected to [EMAIL PROTECTED], port 3306 using table cdr for 7 hours, 41 minutes, 21 seconds. Wrote 70 records since last restart. If you get no such command then that means the module isn't loaded. If you are using localhost then you need to uncomment the sock and make sure it's path is correct. Don't listen to those other guys. The first two were both wrong. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN ASTERISK Cabling...
Alainn wrote: 4) I need to connect 8 isdn phones However, what would I need in order to connect asterisk to ISDN phones within the office? Quad or Octobri cards from junghanns et al. I need not ask that we are operating on a shoestring... The quads will set you back ~600 EUR, the Octobri ~1000 EUR. Maybe you should think about VoIP hardphones ... -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Music on Hold: CPU Intensive Monster
OK. So I did a test last night. All of asterisk's threads where using 0.0% CPU. I made 1 call to our call queue. CPU jumped to average of 9% and stayed around that for the 2 minutes I was in the queue just listening to music on hold. MOH is in MP3 format and I'm using format_mp3. Phone was linksys PAP2-NA using G729. Can I reasonably assume that the 9% was decoding the MP3, then encoding G729? I tried using Anthm's RAW format but that actually made things worse. I tried going back to mpeg321 and asterisk still used the same amount of CPU. Any ideas for getting processor usage down on MOH? -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql does not write to mysql db - SOLVED
i reinstalled the addons and the module works fine now. Thanks to all!! Neal Lawson wrote: using localhost in you mysql conf should work, make sure you linux box and the loopback interface up and has a a entry in your /etc/ hosts for localhost and that your firewall (if you have one setup on your localbox) allows traffic from 172.0.0.1 to 172.0.0.1 On Jul 27, 2005, at 6:17 AM, Dpto. Técnico. wrote: Try to put the IP of you CDR server instead of 'localhost', that's work for me. Regards. - Original Message - From: Kib Eki [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 27, 2005 9:44 AM Subject: [Asterisk-Users] cdr_mysql does not write to mysql db Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_mysql does not write to mysql db
On Wed, 2005-07-27 at 09:44 +0200, Kib Eki wrote: Hi, I configured cdr_mysql (addons 1.0.9) to write the cdr records to the mysql db. The problem is that no records are written to the db. Why? I can import the csv-file to the db. so i assume the db is setup correct. Is there any chance to get debug from cdr_mysql to find his problem? This is my cdr_mysql.conf file: [global] hostname=localhost dbname=cdr password=passw0rd user=root ;port=3306 ;sock=/tmp/mysql.sock userfield=1 Thanks and Regards I do not know if this has changed but I remember that when I first installed the mysql cdr addon the database had to be named asteriskcdrdb and the table where everything is written is cdr. Apart from that make sure that the module is loaded when you start Asterisk. Here is my cdr_mysql.conf: [global] hostname=localhost dbname=asteriskcdrdb table=cdr password=dbpassword user=dbuser ;port=3306 sock=/var/lib/mysql/mysql.sock userfield=1 I can see you are missing the table= from your config. -- Telecomunicaciones Abiertas de Mexico Carlos Chavez Director de Tecnologia +52-55-91169161 Ext. 2001 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem
Thanks for the heads up. I just followed [EMAIL PROTECTED] handbook. Walid -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, July 27, 2005 5:50 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem Hi On Tue, Jul 26, 2005 at 09:09:27PM +0200, Walid Azab wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Neil Cherry Sent: Tuesday, July 26, 2005 6:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 7960 SIP Firmware Upgrade Strange Problem Walid Azab wrote: Thanks to all of you guys. I managed to fix it. It turned out to be that the ZIP file has to be extracted inside the TFTP root not outside then copied to the TFTP root. It is working now. Walid, you should be able to unzip it anywhere and copy it into the directory. It sounds like a permissions problem when you copied it. In the future just make sure that files copied into the tftp directory have at least read permission for everyone (user, group and other). Since it's working now you don't need to fool with it. Just information for the future. Yep you are right , I usually do a chmod 777. 755 would have been enough. 777 allows everyone who happen to get access to your network to change that firmware using simply tftp. Anyone feels like trojaning cisco phones? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Playtones not passing sound to incoming SIP connection
Hi everyone,
I'm in the very early stages of rolling out an asterisk box at work, and one
of the things I'm setting up is a trap for telemarketers ;)
What I want to do is have a sipgate number in the UK here which rings for 10
seconds without calling a hard or softphone, then goes to a voicemailbox.
The problem I'm having is that Playtones doesn't seem to be sending any
sound to the incoming SIP connection. I have added the following to my
[incoming_sipgate] context (which has two other sipgate numbers in there
which both work for incoming and outgoing calls), and on the console I can
see all the lines being executed.
exten = SIPGATEID,1,Wait,1
exten = SIPGATEID,2,NoOp(--- ${CALLERID} calling on Telemarket Divert
Sipgate (${EXTEN}) ---)
exten = SIPGATEID,3,Answer
exten = SIPGATEID,4,Playtones,ring
exten = SIPGATEID,5,Wait,10
exten = SIPGATEID,6,StopPlaytones
exten = SIPGATEID,7,Voicemail(666)
exten = SIPGATEID,8,Hangup
When dialing in via the PSTN number, or from a remote SIP softphone however,
the ten seconds which whould be the Playtones is silence. When the voicemail
kicks in, I can hear the announcements, and leave a message, so I don't
think it's a ports problem. For the playtones line, I have also tried exten
= SIPGATEID,5,Playtones(ring) but doesn't seem to make any difference.
Internally, I set up an extension (in my [default] context - I should have
an [internal] one, I know, ;-P ) with the same commands:
exten = 6613,1,Wait,1
exten = 6613,3,Answer
exten = 6613,4,Playtones,ring
exten = 6613,5,Wait,10
exten = 6613,6,StopPlaytones
exten = 6613,7,Voicemail(666)
exten = 6613,8,Hangup
And when dialing 6613, I get ten seconds of ringtone, then to the
answerphone as expected. There is a difference in the two, in that the
sipgate one has the NoOp line in, but I initially tried the 6613 extension
with that line in (I removed it for ease of differentiation in the console).
Anyone got any ideas on this one? The fact that I can hear the voicemail
announement and leave a message has really thrown me. Maybe I'll just have
to create a .gsm recording of the ringer and use a Playback instead.
Cheers in advance,
Mat
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Re: [Asterisk-Users] Attended transfer not working (atxfer)
When u trasfer u need: Trasfer key sequence trasferee numer talk trasfer key sequence Bruno. Damian Minkov wrote: While on conversation with another party, I dial the atxfer key sequence. Asterisk says Transfer then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep' (language 'en') And then in the console of asterisk is wrote : -- Executing Hangup(Transfered/SIP/8008-432aZOMBIE, ) in new stack Number SIP/8008 is the first originator of the call which must be connected to the transferee. Any ideas? I use CVS of asterisk from 2005-06-16 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- BRUNO DE LUCA Tel. +39 02 9350 4780 (102) FGA Software 20017 Rho - Via Puccini, 8 E-Mail : [EMAIL PROTECTED] Internet: http://www.fgasoftware.com CONFIDENTIALITY NOTICE This message and its attachments are addressed solely to the persons above and may contain confidential information. If you have received the message in error, be informed that any use of the content hereof is prohibited. Please return it immediately to the sender and delete the message. Thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice mailbox on the fly?
On Wed, Jul 27, 2005 at 08:34:11AM -0400, Jim Archer wrote: Thanks for the reply... Well I need the voice mail WAV file mailed to a different email address, depending upon what the code is. Send them all to the same local address and use a simple procmail filter to route them onwards. Or use your MTA's favorite filtering method. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] H323 Configuration file
Hi, This is what I have and is working just fine. I disabled Asterisk gatekeeper and registered directly to a Cisco CallManager 3.3.4 via h323 trunk. ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Configure TCP port range to be used by H.323 ; tcpStart=1 tcpEnd=2 ; ; Configure UDP port range to be used by H.323 ; Note: The port range used by RTP are configured from ; rtp.conf ; udpStart=1 udpEnd=2 ; ; Enable fast start (yes,no). ; fastStart=no ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=no ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=none ; ; Set the maximum number of inbound/outbound/simultaneous ; H.323 connections. ; outboundMax=10 inboundMax=10 simultaneousMax=10 ; ; Set the bandwidth limit for H.323 connections. ; The value is in Kbps. ; bandwidthLimit=1024 ; ; Set tracing options for the wrapper library and for the ; OpenH323 library. ; libTraceFile can be 'stdout' or a full path name to the tracefile. ; Only trace info for OpenH323 is logged in libTraceFile. ; wrapLibTraceLevel=3 libTraceLevel=3 libTraceFile=/root/h323.log ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; ;gatekeeper=192.168.2.2 gatekeeper=DISABLE AllowGKRouted=yes ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the gatekeeper registration timeout ; gatekeeperTTL=600 ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=TONE ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; ;context=voip-h323 context=from-pstn ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; alias=asterisk alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; context=all-aliases alias=ASTERISK alias=666 ; ; Aliases/prefixes routed in more-aliases context. ; context=more-aliases alias=665 ; ; Aliases/prefixes routed in all-prefixes context. ; context=all-prefixes gwprefix=00 gwprefix=01 ; ; Aliases/prefixes routed in more-stuff context. ; ;context=from-pstn ;alias=fax ;gwprefix=14002 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G726- G.726(32k) ; G72616K - G.726(16k) ; G72624K - G.726(24k) ; G72632K - G.726(32k) ; G72640K - G.726(40k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; codec=G711A frames=20 ;codec=G711U ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=2 ;codec=G729 ;frames=2 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, July 27, 2005 4:40 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] H323 Configuration file Folks! I would appreciate if someone could send me a simple working h323 configuration file oh323.conf that is part of [EMAIL PROTECTED] installation. I have tried to use the oh323.conf content listed on WIKI but it is
[Asterisk-Users] does not implement 'PUBLISH'
Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap'
Is udev working properly? I had some issues with that a while back.. I had to hack the udev scripts to get it to create /dev/zap correctly. On Wed, 2005-07-27 at 05:30, Lee Archer wrote: I tend to make it pause for 10 secs when loading the module as I have had a few occurances of loading before /dev/zap has been populated. Wouldn't trust the 2.6.12 kernel as far as I could virutally throw it. Has anyone had any problems with PCI-X systems? In particular call dropping? Regards Lee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Altus Snyman Sent: 27 July 2005 12:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap' I just did the modprobe 2 times and it worked but that was on the 2.6.9 kernel Something about core 3 taking its time to create the device modprobe zaptel sleep 3 modprobe zaptel :-) Peter Raaijmakers wrote: Hi, In struggeling with this problem for a two weeks now. I have a X100P clone card in my * box but I'm not able to get it to run. I'm running fedora core 3 with kernel 2.6.12-1.1372_FC3 on a VIA EPIAML500EA The compiling of both zaptel and asterisk went without any errors. I can run zaptel and asterisk without any errors. When I run ztcfg I don't get any errors too. But when I try to place a call trough my x100p I get this error message in asterisk: NOTICE app_dial.c:1091 dial_exec_full: Unable to create channel op type 'Zap' Outside calls are not comming in either. Here are my zapata.conf and zaptel.conf: -zapata.conf- [channels] signalling=fxs_ks context=incoming channel=1 -zaptel.conf- loadzone = nl defaultzone=nl fxsks=1 --- The funny part comes here: I'm installing a *box for a friend with a ISDN card and the same problem occures. So I probarbly doing something wrong in fedora... Any ideas??? Thanks, Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.338 / Virus Database: 267.9.5/58 - Release Date: 25/07/2005 -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp
opencall.org seems to be off air since yesterday. I am wondering if anyone has a private cache of the most current spandsp they would be willing to share... regards, -dorn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial through IAX to FWD
Hi.. I am trying to do something but it is giving me some hard time here. I have an IAX2 trunk to FWD which is registered and working just fine. I have= 011|. as my dial pattern to allow that. But if I want to dial a toll free number I would have to dial 011*1800XXX What trunk dial rule should I use to enable anyone to call a toll free number by simply dialing 1800XX instead of having to dial 011*1800XXX? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Registration failed problems/Polycom 500/maybe nat problem?
On Tue, 2005-07-26 at 17:53 -0400, Chris Mason (Lists) wrote: I have sent a Polycom IP500 phone to an overseas remote user who has a Speedtouch adsl modem/router. The phone is connected to the 4 port router which performs NAT for the network. The phone rboots, find the server, downloads it's config and tried to register. Then it immediately becomes UNREACHABLE. Jul 26 17:44:41 NOTICE[21026]: chan_sip.c:9138 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '209.59.xxx.xxx' 816/816209.59.xxx.xxx D N 255.255.255.255 10979UNREACHABLE I have tried everything I can think of. Is it possible the ISP is blocking the SIP packets going back to the phone? Chris, I don't know if the ISP is messing with your SIP packets but I can share my experiences with Alcatel SpeedTouch 510v4 ADSL modems SIP and they are not positive with firmware 4.2.7 and older. There is a new firmware version (4.3.1) that has less braindead NAPT abilities. So far I can register my Asterisk box with sipgate.de sipgate.co.uk fine. Perhaps you could try to upgrade the firmware of the modem. An interesting new feature is that 4.3.1 allows you to assign the public IP address of your connection to a LAN device after going through the NAPT rules you have entered. Pasted from the modem's config webpage: This page allows you to assign the public IP address of your Internet Connection(s) to a specific device on your local network... You might want to do this if: * You encounter issues with some applications through the Network Address Translation engine of your SpeedTouch. * This device is running server applications (web server, ...) and you want it to be accessible from the internet. * This device has to be considered as the unique entry to your local network (DMZ). Try with ordinary NAPT rules first and if that doesn't work perhaps try to assign the ADSL IP address to the phone (I don't know about the security implications though). Good luck! Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel error: Unable to create channel op type 'Zap'
what does zttool says ? output of `ls /proc/zaptel/' ?? did you tried ztcfg -vv for verbose output? lsmod show the correct modules for your device? does lspci show the PCI cards? best regards On 7/27/05, Altus Snyman [EMAIL PROTECTED] wrote: I just did the modprobe 2 times and it worked but that was on the 2.6.9 kernel Something about core 3 taking its time to create the device modprobe zaptel sleep 3 modprobe zaptel :-) Peter Raaijmakers wrote: Hi, In struggeling with this problem for a two weeks now. I have a X100P clone card in my * box but I'm not able to get it to run. I'm running fedora core 3 with kernel 2.6.12-1.1372_FC3 on a VIA EPIAML500EA The compiling of both zaptel and asterisk went without any errors. I can run zaptel and asterisk without any errors. When I run ztcfg I don't get any errors too. But when I try to place a call trough my x100p I get this error message in asterisk: NOTICE app_dial.c:1091 dial_exec_full: Unable to create channel op type 'Zap' Outside calls are not comming in either. Here are my zapata.conf and zaptel.conf: -zapata.conf- [channels] signalling=fxs_ks context=incoming channel=1 -zaptel.conf- loadzone = nl defaultzone=nl fxsks=1 --- The funny part comes here: I'm installing a *box for a friend with a ISDN card and the same problem occures. So I probarbly doing something wrong in fedora... Any ideas??? Thanks, Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recording suddenly stopped
Hi..
I noticed all
recording activities suddenly stopped. It seems as if Asterisk is unable to
manipulate files. Here is a sample of a session in which I dialed the Voice Mail
system and tried to record my name:
Any
ideas?
Thanks
Executing
VoiceMail("SIP/100-69a9", "[EMAIL PROTECTED]") in
new stack -- Playing 'vm-theperson' (language
'en') -- Playing 'digits/1' (language
'en') -- Playing 'digits/0' (language
'en') -- Playing 'digits/0' (language
'en') -- Executing VoiceMailMain("SIP/100-69a9", "[EMAIL PROTECTED]") in new stack
-- Playing 'vm-password' (language 'en') -- Playing
'vm-youhave' (language 'en') -- Playing 'vm-no' (language
'en') -- Playing 'vm-messages' (language
'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-starmain' (language
'en') -- Playing 'vm-opts' (language
'en') -- Playing 'vm-helpexit' (language
'en') -- Playing 'vm-options' (language
'en') -- Recording the message --
Playing 'vm-rec-name' (language 'en') -- Playing 'beep'
(language 'en') -- x=0, open writing:
voicemail/default/100/greet format: wav49, (nil)
-- Playing 'vm-review' (language 'en') -- Executing
Macro("SIP/100-69a9", "hangupcall") in new stack --
Executing ResetCDR("SIP/100-69a9", "w") in new stack --
Executing NoCDR("SIP/100-69a9", "") in new stack --
Executing Wait("SIP/100-69a9", "5") in new
stack
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Re: [Asterisk-Users] cdr_mysql does not write to mysql db
Neal Lawson wrote: using localhost in you mysql conf should work, make sure you linux box and the loopback interface up and has a a entry in your /etc/ hosts for localhost and that your firewall (if you have one setup on your localbox) allows traffic from 172.0.0.1 to 172.0.0.1 Don't you mean 127.0.0.1 ? Plus, in the MySQL API documentation the use of localhost indicates to the API that you want to use a socket connection. If you don't want to use sockets (which you should on local machine), the you need to change to an IP to use TCP. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold: CPU Intensive Monster
Matthew Boehm wrote: Any ideas for getting processor usage down on MOH? Encode it in the format you want to send out would help a little. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Klicking sounds in background
Just a thought: do you have DSL on the PSTN line and are you using a line filter? MARK. Jochen Witte wrote: Hello, I have an IVR with Intel HMP SIP stack, which is a peer behind my Asterisk box (Asterisk 1.0.7, Digium PRI). When dialing in via PSTN, there are klicking sounds in the background, which do not appear, when dialing in via SIP (using Asterisk as pbx). The issue does not seem to be an alaw/µlaw problem. I tried trunking two Asterisk boxes via IAX and then call via two asterisks, but the same effect appears. Whenever there is PSTN involved, I have these klicking sounds, when there is no PSTN, everything works correctly. The setup works great with different SIP peers (others than the Intel...) Anyone has an idea? Best regards Jochen -- Jochen Witte email: [EMAIL PROTECTED] web: http://alpha-lab.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] does not implement 'PUBLISH'
On Wed, 2005-07-27 at 11:22 -0400, Hall, Eric M. wrote: Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G Have a look at the very long thread yesterday on this very subject. And then update from CVS. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: R: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap'
why not just compile the kernel from source?? compiling the 2.6.x kernels are easy well.. of course FC is based on deadrat too... maybe switch to slackware... On Wed, 2005-07-27 at 06:01, Yousef Herzallah wrote: I got the same problem yesterday and i just ch'ange the kernel from 2.6.12-1.1372_FC3 to 2.6.9 kernel of fedora core 3, and now it work perfectly. I think there is some problems with the new kernel of fedora. -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Altus Snyman Inviato: mercoledì 27 luglio 2005 13.45 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: Re: [Asterisk-Users] Zaptel error: Unable to create channel op type'Zap' I just did the modprobe 2 times and it worked but that was on the 2.6.9 kernel Something about core 3 taking its time to create the device modprobe zaptel sleep 3 modprobe zaptel :-) Peter Raaijmakers wrote: Hi, In struggeling with this problem for a two weeks now. I have a X100P clone card in my * box but I'm not able to get it to run. I'm running fedora core 3 with kernel 2.6.12-1.1372_FC3 on a VIA EPIAML500EA The compiling of both zaptel and asterisk went without any errors. I can run zaptel and asterisk without any errors. When I run ztcfg I don't get any errors too. But when I try to place a call trough my x100p I get this error message in asterisk: NOTICE app_dial.c:1091 dial_exec_full: Unable to create channel op type 'Zap' Outside calls are not comming in either. Here are my zapata.conf and zaptel.conf: -zapata.conf- [channels] signalling=fxs_ks context=incoming channel=1 -zaptel.conf- loadzone = nl defaultzone=nl fxsks=1 --- The funny part comes here: I'm installing a *box for a friend with a ISDN card and the same problem occures. So I probarbly doing something wrong in fedora... Any ideas??? Thanks, Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] voicemail ODBC storage question
Hello guys. Did anybody use voicemail ODBC storage feature? All voicemessages fields name are clear except for dir, which is assigned to /var/spool/asterisk/voicemail/default/1234/INBOX. Why do we need any directory when we store voicemessages in the database? Noreover, that field is choosen as the KEY or INDEX (marked as MUL in README.odbcsrorage, so I coulnd't say definitely what's that). Why? Much thanks. -- Best regards, Timur Elzhov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why is sip saying NO NAT
nat=1 ? On Wed, 2005-07-27 at 01:43, Chris Mason (Lists) wrote: In the sip conversation below, the traffic is sent to the client and Asterisk is saying noNAT However, this client is configured nat=yes qualify=yes canreinvite=no So why does it still say noNAT? The IP of the phone is 10.0.0.xx Retransmitting #4 (no NAT) to 209.59.xx.xx:10979: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 207.44.xx.xx::5060;branch=z9hG4bK3071ac05 From: asterisk sip:[EMAIL PROTECTED];tag=as121afa0c To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Date: Wed, 27 Jul 2005 08:38:07 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Length: 0 -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP ATA's as house phones
Title: SIP ATA's as house phones I've done a bit of searching and haven't found any good reference for this, so I figured I would ask the group next. Being new to Asterisk, we've been sucessful in figuring out most configuration items, however we're stuck on one thing... Using recent Asterisk stable, we have a situation where I have an analog door phone which, when the button is pushed, simply off hooks the line (no auto-dial) and acts like you just connected tip and ring. In our previous install we used an Inter-Tel Access which we simply created a house phone setup on that analog card extension which rang to an ACD group that hunted various people who were logged in. Is there an equivilent way to do this with Asterisk and a SIP ATA? We're trying to use a Sipura 1001 ATA with our Viking door phone (model info here: http://www.vikingelectronics.com/products/view_product.php?pid=104). Any suggestions? Best Regards, Colin Stefani ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] does not implement 'PUBLISH'
Hall, Eric M. wrote: Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' That was a bug from yesterday's CVS head that is now fixed. Please update! /O ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] does not implement 'PUBLISH'
This bug has been fixed... update to the latest CVS... Regards, Derek - Original Message - From: Hall, Eric M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, July 27, 2005 9:22 AM Subject: [Asterisk-Users] does not implement 'PUBLISH' Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Random Behavior on Trunk Lines with TDM Card
We have implemented * in one of our branch offices and recently ran up against a very strange issue. On random occasions, when we would dial out using our trunk lines, we would get a message stating you do not have to dial a 1 or 0 when calling this number even if we didnt dial a 1 or 0 in the dial sequence at all. After much troubleshooting, we found users with similar issues that simply put a wait (w$EXTEN) in the dial sequence and fixed the problems. We did the same and it automagically fixed itself. This is great, of course, but I was wondering if someone out there could give me a more detailed explanation of why this works and why you have to implement this trick with some telecom providers and not with others. Verison provides our trunks at our branch office, but this issue has never arisen at our home office where we have SBC. We are running * v1.0 with a Digium 4 port FXO TDM card. Thanks, Heath Bowlin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording suddenly stopped
On Wednesday 27 July 2005 12:24, Walid Azab wrote: I noticed all recording activities suddenly stopped. It seems as if Asterisk is unable to manipulate files. Here is a sample of a session in which I dialed the Voice Mail system and tried to record my name: What's the output of df -h? Are you running into a disk quota? Did filesystem permissions change? You need to run some basic basic diagnostics before posting to the list, or at least give details which indicate that you've performed these steps. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] does not implement 'PUBLISH'
Hall, Eric M. wrote: Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G It was a bug in CVS-HEAD, that got fixed yesterday. Upgrade and you'll be fine. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX over HTTP
On 27 Jul 2005, at 09:53, James Cloos wrote: Rob == Rob Scott [EMAIL PROTECTED] writes: Rob For work environments where you only get HTTP or HTTPS access, Rob what is the feasibility of doing IAX over HTTP? Tunnelling tcp/ip over http/(tls/)?tcp/ip is viable, tunnelling rtp/udb/ip or iax/udp/ip over http/(tls/)?tcp/ip however will only work reliably if the tcp doesn't see any packet loss. Else it will retransmit lost packets and the voice quality will suck. That said, if you can get a http or https socket going you can probably also tunnel over dns. So you may want to look into ip over dns/udp/ip tunnels for rtp or iax. -JimC -- James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com It strikes me that if the apps creating each end of the tunnel were IAX aware it would probably be possible to mask most of the bad effects of the TCP channel. Provided that there was enough bandwidth and the latency was low it might be possible to do quite a good job over TLS. TLS ensures that you keep the packet boundaries, which raw TCP wouldn't. That all pre-supposes that there are no HTTP(S) proxies/accelerators/ concentrators in the path! If someone puts a big enough bounty on it I'll have a go - but I have no need of it myself - so it would need to be pretty big ! Tim. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp
dorn hetzel wrote: opencall.org seems to be off air since yesterday. I am wondering if anyone has a private cache of the most current spandsp they would be willing to share... Dorn, spandsp is on www.soft-switch.org Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom gain settings
Hi All, I have some Polycom IP300's and I'm interested in increasing the max volume for the headset (not handset), I'm wondering if anyone has experience adjusting these values: gains voice.gain.rx.analog.handset=0 voice.gain.rx.analog.headset=0 voice.gain.rx.analog.chassis=3 voice.gain.rx.analog.chassis.obs=-12 voice.gain.rx.analog.chassis.IP300=-6 voice.gain.rx.analog.ringer=3 voice.gain.rx.analog.ringer.IP300=-6 voice.gain.rx.digital.handset=0 voice.gain.rx.digital.headset=-21 voice.gain.rx.digital.chassis=0 voice.gain.rx.digital.chassis.obs=0 voice.gain.rx.digital.ringer=-21 voice.gain.rx.analog.handset.sidetone=-24 voice.gain.rx.analog.headset.sidetone=-24 voice.gain.tx.analog.handset=3 voice.gain.tx.analog.headset=3 voice.gain.tx.analog.chassis=3 voice.gain.tx.analog.chassis.obs=6 voice.gain.tx.digital.handset=0 voice.gain.tx.digital.headset=0 voice.gain.tx.digital.chassis=6 voice.gain.tx.digital.chassis.obs=4 voice.gain.tx.analog.preamp.handset=23 voice.gain.tx.analog.preamp.headset=23 voice.gain.tx.analog.preamp.chassis=42 voice.gain.tx.analog.preamp.chassis.obs=20 / Thanks, Pete ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] does not implement 'PUBLISH'
Got it! Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Wednesday, July 27, 2005 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] does not implement 'PUBLISH' On Wed, 2005-07-27 at 11:22 -0400, Hall, Eric M. wrote: Not sure what this is. When I call my own ext the call will ring for 10 sec and goto the voicemail. However the phone will keep ringing and I see this on the asterisk CLI Jul 27 11:17:48 WARNING[3563]: chan_sip.c:8666 handle_response: Host '192.168.0.200' does not implement 'PUBLISH' Have no idea what this is talking about 192.168.0.200 is a cisco 7960G Have a look at the very long thread yesterday on this very subject. And then update from CVS. -- Dave Cotton [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sending DTMF Tones Offhook
Greetings All! The Asterisk Call Manager works great. But I have one question for anyone who has used it. I cannot get the system to send some DTMF tones down the channel once the call has been made. Below is the script I am using to make the call, and start recording the channel. I am starting to make a system the will use asterisk to become an automatic random quality monitoring system that will dial into an Aspect (or any for that matter) ACD that has CTI and do it's job. The ZapDialOffhook event says it's successful in the API window but does not seem to do any dialing (i hear no tones). Is there another command that works? I have tried using the CLI command SendDTMF to no avail. Everything below works except the -zapdialoffhook-. Thanks in advance for any assistance you wish to provide! Connect --- telnet 172.18.128.74 5038 login -- action: login username: *** secret: *** make call: -- action: originate channel: zap/1/98744207 context: default exten: 5196 priority: 1 callerid: test async: yes dial digits: -- (try#1) Action: ZapDialOffhook ZapChannel: 1-1 Number: 12334556 (try#2) Action: ZapDialOffhook ZapChannel: 1 Number: 12334556 (both do the same thing) record call: (zap) -- Action: Monitor Channel: Zap/1-1 File: recordingtest14 Mix: 1 stop recording call: -- Action: StopMonitor Channel: Zap/1-1 end call: -- action: hangup channel: zap/1-1 -Aaron Sundman www.nerdpc.com www.grcu.net ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Random Behavior on Trunk Lines with TDM Card
On Wednesday 27 July 2005 11:51, Heath Bowlin wrote:
We have implemented * in one of our branch offices and recently ran up
against a very strange issue. On random occasions, when we would dial
out using our trunk lines, we would get a message stating you do not
have to dial a 1 or 0 when calling this number even if we didn't dial a
1 or 0 in the dial sequence at all. After much troubleshooting, we found
users with similar issues that simply put a wait (w$EXTEN) in the dial
sequence and fixed the problems. We did the same and it automagically
fixed itself. This is great, of course, but I was wondering if someone
out there could give me a more detailed explanation of why this works
and why you have to implement this trick with some telecom providers and
not with others. Verison provides our trunks at our branch office, but
this issue has never arisen at our home office where we have SBC. We are
running * v1.0 with a Digium 4 port FXO TDM card.
I am willing to bet that you dialled a number with '1' or '0' as the second or
even third digit of the full number, and that your LEC is either overloaded
and/or running older, slower technology.
What I believe is happenning is that Asterisk is picking up the line and
dialing, but the LEC switch isn't fast enough and only begins listening for
DTMF after Asterisk's already played the first few digits. This is well
documented and the typical solution is to Dial(Zap/g1/w${EXTEN}) instead of
just Dial(Zap/g1/${EXTEN}).
The problem stems from a combination of the LEC switch being slow to begin
detecting DTMF and from the Zaptel drivers not waiting for dialtone. You may
notice that this problem only occurs at certain times of the day and this is
due to the LEC's call volume. When you call during busy times it may take a
few seconds for the LEC's switch to get around to your line. Do you
sometimes notice that you don't hear dialtone (with a normal phone hooked up
to the line) for a second or two?
-A.
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Re: [Asterisk-Users] Dial through IAX to FWD
On Wed, 27 Jul 2005 18:07:23 +0200 Walid Azab [EMAIL PROTECTED] wrote: Hi.. I am trying to do something but it is giving me some hard time here. I have an IAX2 trunk to FWD which is registered and working just fine. I have = 011|. as my dial pattern to allow that. But if I want to dial a toll free number I would have to dial 011*1800XXX What trunk dial rule should I use to enable anyone to call a toll free number by simply dialing 1800XX instead of having to dial 011*1800XXX? Thanks Are you using [EMAIL PROTECTED] or setting up the configs yourself?? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp
On Wed, Jul 27, 2005 at 11:17:28AM -0400, dorn hetzel wrote: opencall.org seems to be off air since yesterday. I am wondering if anyone has a private cache of the most current spandsp they would be willing to share... It is now actually http://soft-switch.org/ . It does respond, though generally slow. Anyway, grab the sources of the packages from http://updates.xorcom.com/iso : http://updates.xorcom.com/rapid/pool/xorcom/asterisk-spandsp-plugins_0.0.20050612.orig.tar.gz http://updates.xorcom.com/rapid/pool/xorcom/spandsp_0.0.2pre18.orig.tar.gz -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
