[asterisk-users] H263-2000 video format
I'm trying to connect my asterisk 1.4.6 to a system that provides video content (through SIP). Problem is my video system only speaks H263-2000 version (aka H263++). As far as I can see, * only understands H263 and H263+ in and sdp. Can anybody tell me how to extend asterisk so it'll support H263++? From what I've heard, H263+ and H263++ should be compatible. So I was thinking there's no need for a new codec. Am I right? Cheers, K ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] gtalk - no audio
Hi list, I'm trying to get channel gtalk working in asterisk 1.4.5 I have it built and configured as follows: *jabber.conf:* [general] debug=yes autoprune=no autoregister=no [myaccount] type=client serverhost=talk.google.com [EMAIL PROTECTED]/Talk secret=mypassword port=5222 usetls=yes usesasl=yes statusmessage=Talk to me timeout=100 *gtalk.conf:* [general] context=default allowguest=yes bindaddr=172.25.123.18 [guest] disallow=all allow=ulaw context=gtalk This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. I first had similar problems, because I didn't use bindaddr in gtalk.conf. But that fixed it for me, but not for most other cases. Also, we all use the same network (same routing and NAT) and Gtalk version. Audio calls between regular Gtalk users is not a problem. This problem really puzzles me. Is it a channel gtalk problem, or do we need to look at other settings (network, client settings...)? I personnaly think we can rule out network config, since both successful and unsuccessful users work in the same lan. Is there anybody with experience in using channel gtalk? Should we start debugging? What can we learn from jabber debug logs? Any help is very much appreciated! koenvi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gtalk - no audio
Would you be so kind to share your experience? I can read most of C language, but writing it is another thing. And I'm not familiar with the internals of Asterisk... Or maybe you could already confirm that my problem is related to NAT (client or Asterisk side, not sure) On 6/21/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I been to this scenario before. But I got mine working just last May 2007 and it appears to be stable now ready for some serious commercial application. Hint: if you have any experience with C, try to check with the source code related to the channels you are stressing down here. Well, it is not an easy task to do and not for the faint of heart. With a little luck and more of motivated creativity, you will get it working. Trust me, been there done that. Hi list, I'm trying to get channel gtalk working in asterisk 1.4.5 I have it built and configured as follows: *jabber.conf:* [general] debug=yes autoprune=no autoregister=no [myaccount] type=client serverhost=talk.google.com [EMAIL PROTECTED]/Talk secret=mypassword port=5222 usetls=yes usesasl=yes statusmessage=Talk to me timeout=100 *gtalk.conf:* [general] context=default allowguest=yes bindaddr=172.25.123.18 [guest] disallow=all allow=ulaw context=gtalk This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. I first had similar problems, because I didn't use bindaddr in gtalk.conf. But that fixed it for me, but not for most other cases. Also, we all use the same network (same routing and NAT) and Gtalk version. Audio calls between regular Gtalk users is not a problem. This problem really puzzles me. Is it a channel gtalk problem, or do we need to look at other settings (network, client settings...)? I personnaly think we can rule out network config, since both successful and unsuccessful users work in the same lan. Is there anybody with experience in using channel gtalk? Should we start debugging? What can we learn from jabber debug logs? Any help is very much appreciated! koenvi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gtalk - no audio
I haven't changed rtp.conf from original installation. So the values are: rtpstart=1 rtpend=2 I should maybe give it a try with a lower rtpstart. What do you mean by turning on NAT? Are you referring to parameter bindaddr in gtalk.conf? (found that on http://www.voip-info.org/wiki/view/Asterisk+Google+Talk) Thanks already! On 6/21/07, Joseph Bajin [EMAIL PROTECTED] wrote: what does your RTP settings look like? I had problems with this at first. One thing I made sure of was that NAT was turned on and that the rtpstart in the rtp.conf file was set to 2000 and the rtpend was up to 2 (but you can make that much higher). Gtalk seems to have a very low RTP port that it uses for media. On 6/21/07, Philippe Sultan [EMAIL PROTECTED] wrote: Hi Koen This works fine when I call this account from my personal gtalk. But others have some very strange problems. In most cases, I see the call coming into Asterisk and executing normally. On the callers side, the call looks like it was answered, but there's no audio. In some other cases, the call doesn't even appear to be answered, although I see a normal execution on Asterisk. Can you please open a bug report that describes your problem, and attach an Asterisk debug output for a failed call to the report? Thanks, Philippe ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and 3PCC
Gregory, I know there is something called SIP CTI TR87. It's used by Nortel to integrate with Microsoft's Live Communication Server. Don't know if something similar exists for Asterisk. This links could be helpfull: http://www.ecma-international.org/publications/techreports/E-TR-087.htm Regards, Koen On 1/9/07, Gregory Duchatelet [EMAIL PROTECTED] wrote: Hi all, I have an Asterisk server running, and some hardware phones, and I want to do 3PCC : third party call control. The third party is a software running on the asterisk box, which can for example ask a hard SIP phone to put a call on hold. To do that, this software has to send a SIP message to this phone. How can I do that ? Greg ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cut function on semicolon separator
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a semicolon as
separator.
I was thinking the 'CUT' function would be perfect for this.
But the problem is the semicolon. In the dialplan it is always understood as
a separator for parameters.
What I have tried so far:
[macro-eva-on-sip]
exten = s,1,NoOp(${CALLERID(name)})
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(v=${CALLERID(num)})
exten = s,n,Set(sep=;)
exten = s,n,NoOp(${CUT(v,sep,1)})
exten = s,n,Dial(SIP/evavox/${MACRO_EXTEN})
exten = s,n,Hangup()
I'm convinced there's a very simple solution to this, but I don't see it.
Anybody?!
Grtz,
Koen
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Re: [asterisk-users] Cut function on semicolon separator
Peter,
Thanks for your reply!
It didn't work though.
There's actually already a problem setting the semicolon as value for the
'sep' variable.
*The functions:*
exten = s,n,Set(sep=';')
exten = s,n,NoOp(${CUT(v,${sep},1)})
*The output:*
-- Executing Set(SIP/1649-09ca84f0, sep=) in new stack
-- Executing NoOp(SIP/1649-09ca84f0, 1649;phonecontext=Exp_Net) in new
stack
fyi, v is a variable holding 1649;phonecontext=Exp_Net
So the question is now: how can I set a variable to hold a semicolon as
variable.
And can I then use this variable as separator in the Cut function?
On 11/30/06, Peter Lindquist [EMAIL PROTECTED] wrote:
Hi Koen,
Try:
exten = s,n,NoOp(CUT(${v},${sep},1))
Cheers
Koen Van Impe wrote:
Hi,
I have the most stupid problem in my dialplan.
I need to do something as trivial as splitting a string, with a semicolon
as separator.
I was thinking the 'CUT' function would be perfect for this.
But the problem is the semicolon. In the dialplan it is always understood
as a separator for parameters.
What I have tried so far:
[macro-eva-on-sip]
exten = s,1,NoOp(${CALLERID(name)})
exten = s,n,NoOp(${CALLERID(num)})
exten = s,n,Set(v=${CALLERID(num)})
exten = s,n,Set(sep=;)
exten = s,n,NoOp(${CUT(v,sep,1)})
exten = s,n,Dial(SIP/evavox/${MACRO_EXTEN})
exten = s,n,Hangup()
I'm convinced there's a very simple solution to this, but I don't see it.
Anybody?!
Grtz,
Koen
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Re: [asterisk-users] Cut function on semicolon separator
All,
The last Peter got it right! :-)
The final solution:
exten = s,n,Set(sep='\;')
exten = s,n,NoOp(${CUT(v,${sep},1)})
Thanks for you input and have a very nice day!
Koen
On 11/30/06, Peter Boehm [EMAIL PROTECTED] wrote:
_The functions:_
exten = s,n,Set(sep=';')
exten = s,n,NoOp(${CUT(v,${sep},1)})
Have you tried to put a '\' in front of the ';': Set(sep='\;')?
Peter
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Re: [asterisk-users] How to enable jingle in 1.4beta2?
Afer running ./configure with whatever options you need, you should run make menuselect That will give you a menu to select the required modules. Modules marked with XXX are disabled, mostly because of a missing dependency. I think jingle requires iksemel. Good luck! Koen On 9/28/06, Raffaele Porzio [EMAIL PROTECTED] wrote: Hi everyone, I'm going to compile this version of asterisk with jingle support, but I need to know how to enable this feature with the ./configure --enable option the sources chan_jingle.c and res_jabber.c aren't compiled by defalut. Thanks everyone___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Clusters
Sounds like a nice setup you have in mind. All I can tell is that you might have trouble with clocking on your PRI's if you use multiple cards in one system. I've read about it somewhere, but can't find the source. Have a look at the wiki. Syncing clocks on one card happens on the card level. But of you have multiple cards, there might be some slip. Good luck! On 9/7/06, Mitch Thompson [EMAIL PROTECTED] wrote: Hello, All.I've been lurking on this list for some time, trying to drink from thefire hose. Now, I have a few questions. First, though, here is the background:I work for a testing facility where we test telephony products. We havebeen using Asterisk for about 4 months now as a test bed for variousthings. Now, our test engineers want to ramp things up a bit. Essentially, the system under test has the capability of using up to 8T-1 ISDN/PRI lines. The engineers want to build two Asterisk clusters,each with 20 ISDN/PRIs. Of these, 12 would be Inbound PRIs, and the other 8 would be Outbound. The system under test would be connected to8 of the Inbound lines, and a Call generator, such as an AmeritecCrescendo, would be connected to the other 4 Inbounds. These twelve lines would dial through the Asterisk, through the 8 Outbound lines tothe other Asterisk, which would terminate the 8 PRIs into 12 PRIs worthof Crescendo or Fortissimo.The whole purpose of this mess is to determine how the system under test responds to network congestion, since it is competing with the Crescendofor the 8 Outbound PRIs.So, I guess my questions are:1) Is Asterisk's congestion capabilities robust enough to do what we want? 2) I have the resources to build this cluster one of two ways:a) I have 4 Dell PowerEdge SC1600's, with 3.0 GHz Dual Xeons (looks like4 processors to the system), 2 GB of RAM, and 6 slots (2xPCI, 2X PCI-X, 2X PCI-Express). I would use these and put 5 Digium cards in each forthe two Asterisk clusters.b) I also have 12 1U rackmounts, 866 MHz, 1 GB RAM each. If I usedthese, I would put 1 Digium card in each and organize them into two groups of 5 Asterisk servers.For #2, which would be better/easier, a or b?I would appreciate any insights anyone may be able to provide.Mitch Thompson--Nothing is more destructive of respect for the government and the law of the land than passing laws which cannot beenforced. —Albert Einstein___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger
I use logrotate too, because I didn't know of the functionality in Asterisk.
Logrotate works fine for me though.
Kenny, you should give it a try!
K
On 7/28/06, Filip Drągowski [EMAIL PROTECTED] wrote:
asterisk does daily log rotate all along ? i didn't know that it is posiiblei create file in /etc/logrotate.d/asterisk (copy of postgrsql and renamed it)/var/log/asterisk/full {
daily rotate 10 copytruncate delaycompress compress notifempty create 640 root root}i have 10 last full log, 8 oldest gziped
on one cosnole a do asterisk -r
on other i do asterisk -rx logger rotate
and the result is
-- Remote UNIX connection
Asterisk Event Logger restarted
Asterisk Queue Logger restarted
-- Remote UNIX connection disconnected
how often new log files are created ? = how many log files are created
in 1 second ?
Very often, it slows down as the number of log files in the
/var/log/asterisk directory increases.
there is some kind of regularity or it is done randomly (10logs/1s and
another time 20logs/1s) ?
Seems to happen more often than not (the past three days) overnight.
We're assuming that it starts when asterisk does it's daily log rotate
itself and then gets itself into a spin...
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Re: [Asterisk-Users] Receiving faxes and then sending them on
Maye you should use the 'D' option in the Dial application to proceed when the call is answered.
Not sure, and I don't have time to test myself, but give it a try!
K
On 6/16/06, Frederik Fix [EMAIL PROTECTED] wrote:
Hi,I'm trying to setup a system where incoming faxes are received usingSpanDSP and then send on to another (remote) fax machine. The SpanDSP
part is working excellently, however I dont seem to be able to getthe forwarding part to work. Heres what I put into my extensions.conf:exten = s,4,Answer()exten = s,5,Set(FAXFILE=/tmp/fax-${UNIQUEID}.tif)
exten = s,6,Set(EMAILADDR=[EMAIL PROTECTED])exten = s,7,Set(EMAILADDR=${ARG1})exten = s,8,rxfax(${FAXFILE}|debug)exten = s,9,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR} $
{CALLERIDNUM})exten = s,10,Dial(${ARG2})exten = s,11,txfax(${FAXFILE}|caller)exten = s,12,HangupAsterisk does start dialing at priority 10 however as soon as theremote fax hangs up that call gets destroyed as well.
Is there anyway to do something like this?Kind regards,Frederik Fix___--Bandwidth and Colocation provided by Easynews.com
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Re: [Asterisk-Users] d e options in meetme()
We use dynamic conferences with MeetMe. As far as I can tell, the 'e' option is not needed. We use a global var as counter for the conference number. You provide it with the MeetMe command. This way you always know which conference to join. K On 6/16/06, Miles Scruggs [EMAIL PROTECTED] wrote: I'm really confused on how to use these two options together:A while back:JustRumours edited this page:http://www.voip-info.org/wiki-Asterisk+cmd+MeetMeand added a little section about dynamic conferences.the 'e' option is repeated all over the page as the savior of dynamic conferences, maybeI'm just dumb, but can someone tell me if a conference is created withthe e d option, how does one even figure out the conference number so that they can tell other people which conference to join.Also whatrange of numbers is this conference automatically generated from, andcan it be controlled?From what I can tell if you pass 'e' then asterisk will think of it's own favorite magic number (that isn't currently being used) and generatea conference with that number, anyone else that wants to join they needto guess what that number is.Sorry for my weak stab at humorous satire but it seems like something is missing here.Thanks Miles___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling mpg123 under 1.2.9.1 / Ubuntu 6.06
Why still use mpg123? Start using format_mp3 from asterisk-addons and your * will play mp3 by itself... K On 6/13/06, Marc Rohlfing [EMAIL PROTECTED] wrote: Hi,I made the mistake of upgrading both my Linux box (to Ubuntu 6.06) andAsterisk (to 1.2.9.1) at the same time. Now, when trying to compilempg123 - using the tried and true make mpg123 -, the build fails withan errormake[3]: Entering directory `/usr/src/asterisk- 1.2.9.1/mpg123-0.59r'make[3]: *** No rule to make target `\', needed by `mpg123'.Stop.Maybe there's someone out there more versed in Linux who has an ideawhat might have gone wrong. Thanks! Marc___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: asterisk and nortel meredian option 11c
Muhammad, I have been struggling with M1 and * over an E1 for a while myself, but know it's running fine. Here's my d-channel config: ADAN DCH 18 CTYP MSDL CARD 08 PORT 1 DES Asterisk1 USR PRI DCHL 8 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC EURO CNTY BEL PINX_CUST 0 ISDN_MCNT 300 CLID OPT1 PROG NCHG CO_TYPE STD SIDE USR CNEG 1 RLS ID ** RCAP COLP MBGA NO OVLR NO OVLS NO T310 120 INC_T306 0 OUT_T306 0 T200 3 T203 10 N200 3 N201 260 K 7 It's a config where Asterisk is master and Meridian is slave in euroisdn. The zapata.conf that goes with that: #---[trunkgroups][channels]context=incoming-prabusydetect=nousecallerid=yescidsignalling=v23usecallingpres=yescallerid=asreceivedswitchtype=euroisdn signalling=pri_net group=1channel=1-15,17-31 #--- Make sure your [trunkgroups] section is empty! I lost a lot of time on that one myself! Zaptel.conf: #--- span=1,0,0,ccs,hdb3bchan=1-15,17-31dchan=16loadzone=bedefaultzone=be#--- We don't use crc4 here, but you can add it if you wish. Good luck! K On 6/8/06, Muhammad Zeeshan Latif [EMAIL PROTECTED] wrote: Hi I want to connect asterisk 1.0.9 ( kernel 2.6.8-2 debian )with TE110P and Nortel meridian option 11c release 25.40 with NTBK50AA card which is An E1 card. But the main problem is the first stage that no sync occurs the * card never syncs with meridian card I am using euroisdn, ccs , hdb3, crc4 , pri_net on asterisk And I am assuming that meridian is using same as it is connected to Nortel passport mvpe card which is an e1 isdn card and using the same config as astresk but the card never see each other. On the contrary when I connect the asterisk with the Nortel passport mvpe card it does detect the mvpe card but the d chan flaps btwn up and down and the hell of HDLC BAD FCS messages appears on the cli of * . I have also tried yellow alarm on the span but not of any help . Can any one tell me the config of meridian option 11c and asterisk and what I am doing wrong. thanks___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install
Use format_mp3 from asterisk-addons. It will enable your * to play mp3 without the use of an external process... (if I got it right) On 6/8/06, Richard Reina [EMAIL PROTECTED] wrote: Turby,Thanks for your replay, but does this mean that * can't play mp3s? I was hoping not to have convert the MOH mp3s that came with * and are supposed to be the native default that * is supposed to be able to play -- not to mention that I don't have sox installed with whick to convert them. Does this mean that 1.2.7.1 has a bug? If so can someone tell me if I should, and how I would go about reporting it.Thanks again for the reply. turby [EMAIL PROTECTED] wrote: convert the moh sounfile to pcm or sln save the file to /var/lib/asterisk/moh/default set the musiconhold.conf [default]mode=filesdirectory=/var/lib/asterisk/moh/default turby@ www.canistec.com From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED]] On Behalf Of Richard ReinaSent: Wednesday, June 07, 2006 5:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install Thank you very much for your relply. No I did not install mpg123 as the instructions at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.conffor version 1.2 say the mpg123 is no longer needed.| Rurouni Alucard | [EMAIL PROTECTED] wrote: Did you check your mpg123 version ?, asterisk needs a specific version in order to work... - Original Message - From: Richard Reina To: asterisk-users@lists.digium.com Sent: Wednesday, June 07, 2006 6:02 AM Subject: [Asterisk-Users] Music On Hold not working with new 1.2.7.1 install I have followed the instructions provided at: http://voip-info.org/tiki-index.php?page=asterisk+config+musiconhold.confincluding installing asterisk-addons-1.2. I have left musiconhold.conf as is, calm-river et al are fine for now.However, no sound is heard and I get this message from the CLI when accessing MOH: -- Started music on hold, class 'default', on channel 'Zap/19-1'-- Stoped music on hold on Zap/19-1This happens whether it's a parked call or whether I access MOH directly via:exten = 800,1,Answer exten = 800,2,MusicOnHold()Any help would be greatly appreciated.Thank you very much.Richard __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How can I use features without enabling 'call parking'?
Is there a way to use 'application mapping' from features.conf without the built in features (pickup, blind transfer, etc.) nor call parking? I have been trying to comment out everything in features.conf, but my asterisk stills shows the defaults... Koen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Global variables - collision?
If I edit the value of a global variable in my dialplan, could there be a risk of collision between calls? More in details: could a global var be used to build a counter that will be incremented by every call that passes. I think when 2 calls come in almost sumiltaneously, they could both be incrementing and saving the same value... which is bad! Anybody knows how asterisk handles this? K ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Global variables - collision?
Sounds like a reasonable explanation.
But this means that I should limit the incrementing stuff to one line in the dialplan.
This would be bad:
exten = s,1,Set(Chan_Var=${GlobalVar})
exten = s,2,Set(Chan_Var=$[${Chan_Var} + 1])
exten = s,3,Set(GlobalVar=Chan_Var,g)
Better:
exten = s,1,Set(GlobalVar=$[${GlobalVar} + 1])exten = s,2,Set(Chan_Var=${GlobalVar})
Please confirm...
K
On 5/31/06, Filip Drągowski [EMAIL PROTECTED] wrote:
Each variable is specyfied by name and callid Call number 1. Executing Set(SIP/X-2749, DL=0) in new stackCall number 2. Executing Set(SIP/X-9100, DL=0) in new stack
X - sip provider login and there is -number (i think that this number is in HEX)so every local variable have diffrent identityAs You can see Asterisk uses stack so there should be:1. Executing Set(global_VAR) in new stack
2. Executing Set(global_VAR) in new stackNo.1 resolves then next ... ther is no simultaneous operationit's my opinion. Try it and see what is shown in * console.-FD
If I edit the value of a global variable in my dialplan, could there be a risk of collision between calls?
More in details: could a global var be used to build a counter that will be incremented by every call that passes.
I think when 2 calls come in almost sumiltaneously, they could both be incrementing and saving the same value... which is bad!
Anybody knows how asterisk handles this?
K
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Re: [Asterisk-Users] Zap Channels , for round-robin search and call
depending on your zapata.conf file, you should use
exten = _9X.,1,Dial(Zap/r1/${EXTEN:1})
The little 'r' means round robin, starting at the next highest channel than last time.
Have a look in extensions.conf from the samples for more options.
Make sure you have your 4 channels in one group (group=1).
K
On 5/31/06, John Joseph [EMAIL PROTECTED] wrote:
HiI am using a 4FXO , TDM400P cardI am able to call outside , after modifiyingextensions.conf
withexten = _9X.,1,Dial(ZAP/1/${EXTEN:1})using this , I can only dial through one of theport , Actually I want todial outside using round -robinsearchAfter reading the manuals , I have plans to
modified the above line asexten =_9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1})Please let me know wheter the above line ,iscorrect to useI think , it will dial any one of the four
channel which is available Pleasegive your comments on theputtingthe lineexten =_9X.,1,Dial(ZAP/1/${EXTEN:1}|ZAP/2/${EXTEN:1}|ZAP/3/${EXTEN:1}|ZAP/4/${EXTEN:1}) Thanks
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Re: [Asterisk-Users] Asterisk Meridian Tie Line
I'm running pretty much the same config in Belgium. Here's what I use: zaptel.conf: span=1,1,0,ccs,hdb3 # no CRC4 used here bchan=1-15,16-31 dchan=16 zapata.conf: [trunkgroups]trunkgroup = 1,16spanmap = 1,1,1 [channels]context=incoming-priswitchtype=euroisdnpridialplan=nationalprilocaldialplan=nationalsignalling=pri_cpegroup=1channel = 1-15,17-31 Works for me, hope it can help you! On 5/18/06, Steve Totaro [EMAIL PROTECTED] wrote: Andy Kirby wrote: I am new to the group but have searched the doc's FAQ's etc before posting here. We are attempting tie our asterisk server/service to the building's PBX, the building is in the UK and the local PBX is a meridian option 11 installed and mainteined by BT. BT Have installed a NTBK50AA E1-PRI card in the meridian with daughter cards NTBK51AA (D channel) and NTAK20BD (Clocking)I have asked BT to configure the card as a Master (Exchange end) E1 Euro ISDN (Just like a standard ISDN30e)They claim to have done this in line with the model they use to interface to Cisco routers etc.I have installed a Digium TE411P in our server looped back the span 2 port (Gives a green light and OK with same config as span 1) and am using a crossover cable to link the PBX to our server. (We tried a pucker BT cross over cable with exactly the same results as mine and a striahgt through gives us nothing at all, I guess as you might expect) I have configured the Zap span for 1 clocking (Primary) 1 line build out, with the framing etc as CCS, HDB3, CRC4 But they don't appear to want to synchronise/talk to each other. ZTTool claims that the span is up and down more times than a fiddlers elbowand the clocking source is internal.( Might I expect the alarm state to be constant if the framing etc was matched and the clock source to show as external ??) The alarms are cylcling from red to red/yellow and finaly to red/yellow/recover before falling back to red and starting again.I think I may be missing something that is probably blindingly obvious to someone in the know.The BT guy has been very good and is trying to help us get this going but seems rather nonplussed with the terms CRC4, CSS and HDB3... Please can somone help and point me (and I guess by extension the BT guy) in the right direction. Cheers AndyTry your end with every different combination of settings applicable toEuroISDN.I would start by removing the CRC4.Try something, if it doesnt work, put it back and change another setting. s___--Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to set channel to linear mode
I have a TE110P connected in euroisdn as pri-cpe. When I dial out from a sip phone to a number over the pri, I get an error Unable to set channel 1 (index 0) to linear mode On the destination phone, I only get a terrible noise when answering the call. There doesn't seem to be a speech path... Config: libpri 1.2.2 - zaptel 1.2.5 - asterisk 1.2.6 on Fedora Core 4 (kernel 2.6.11) Anybody knows what this is all about? K ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE110P on E1
Hi, I wonder if anyone is using Digium's TE110P card on an E1 connection. I have been try to, but so far it wasn't much of a success. It only works more or less in EuroISDN as PRI CPE. And even that config gives me some trouble with channel negotiation. My current config: zaptel.conf: span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone=be defaultzone=be zapata.conf: [trunkgroups] trunkgroup = 1,16 spanmap = 1,1,1 [channels] context=incoming-pri switchtype=euroisdn pridialplan=national prilocaldialplan=national signalling=pri_cpe group=1 channels = 1-15,17-31 I have tried EuroISDN and QSIG in both NET and CPE, without much success. From traces of D-channel messaging, I think there's a problem with channel negotiation. The information element (IE) involved only shows 5 bytes coming from our PBX. But Asterisk (Zaptel) uses 6 bytes. From pri debug messages on Asterisk, I see that it adds a DS1 Identifier. I haven't seen that in other d-channel traces between other systems. Anyone with experience on this matter??? Regards, Koen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
