RE: [Asterisk-Users] codec preferences
You did read the documentation I presume? It clearly states that you need to have separate G729 licenses to have * talk G729, without them * can only pass thru G729, not transcode it (Zap - G729). Since both your sip phones do G729 * is just performing pass thru. Kind regards, E. Versaevel -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Roy Sigurd Karlsbakk Verzonden: maandag 27 december 2004 21:07 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] codec preferences hi Username : 112 Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (gsm|g729|g726|alaw|ulaw) the above is from SIP SHOW PEER 112, and as it clearly shows, g.729 is preferred before alaw. If I dial this SIP - * - SIP from a phone with G.729 enabled, it uses G.729. However, if I dial from my cell phone - GSM - PSTN - * - SIP, the call uses ALAW, which I thought it shouldn't. I'm using Asterisk CVS-v1-0-12/16/04-04:15:36 with the patch from http://bugs.digium.com/bug_view_page.php?bug_id=0003106 all comments welcome roy ___ 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] Music on Hold
If youre using G.711 make sure youve got silence suppression turned OF, seems that the phone only receives rtp while sending it. (or, do as we did, throw the OptiPoint out of the window J) Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens richard Coco Verzonden: maandag 27 december 2004 15:31 Aan: asterisk-users@lists.digium.com Onderwerp: [Asterisk-Users] Music on Hold Hi all, i'm trying to configure MoH for OptiPoint400std SIP, but it doesn't work. If i try with a softclient moh works fine. Somebody experience with OptiPoint400? Is there additional settings on the Optipoint? Any help would be much appreciated!! thx. Do you Yahoo!? Meet the all-new My Yahoo! Try it today! ___ 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] Asterisk billing solution
Qui, mais je ne parle pas français ;) -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Patrick Verzonden: donderdag 23 december 2004 13:28 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: RE: [Asterisk-Users] Asterisk billing solution On Thu, 2004-12-23 at 00:12 +0100, Thierry wrote: Hi I have something like this but it's in french and it uses teh res_config Best regards Thierry wehr Thierry, If you are willing to share your billing solution with the community, I'm sure there will be people pitching in to translate it from french to english and any other language they feel is important. My french is not that good but I will definitely have a look and help where possible. 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 ___ 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 back when no longer busy
Hello, Im trying to implement a function
available on the PSTN net here, if you dial a number which is busy and you
press 5, you will be called back when the busy party hangs up.
Figuring out if a SIP user is busy isnt to
hard, ${DIALSTATUS} produces a BUSY message, however, how can I implement the
call back?
IE, I dial to extension 712, but that extension is
busy, I dial 5 and hangup. If extension 712 is no longer busy Im called
back and the moment I pickup the extension 712 is dialed again.
Im thinking something like putting a value in
the Asterisk DB if the extension is busy, after that extension hangs up it
should trigger the dial to the originating extension and on pickup it should
dial the other extension J
Kind regards,
E. Versaevel
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RE: [Asterisk-Users] Queueueueuueue position
Anyone any idea's? There are 3 to 5 calls in the queue at that moment, all from different CID, hold time is over a minute. -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Brian Roy Verzonden: vrijdag 17 december 2004 1:52 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Queueueueuueue position On Thu, 16 Dec 2004 15:18:10 +0100, E. Versaevel [EMAIL PROTECTED] wrote: When I call in (with an agent logged in) I get to hear the MOH on the client side, hover no matter how high the hold time is, I NEVER get an announcement over my queue position or my estimated wait time? Both the incoming call and the agent are on SIP channels. What is wrong ? Kind regards, E. Versaevel Would that be because this is the only call in queue? Try putting another call in queue and see what you get. -Chuji ___ 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] Queueueueuueue position
There are 3 to 5 calls in the queue at that moment, all from different CID, hold time is over a minute. -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Brian Roy Verzonden: vrijdag 17 december 2004 1:52 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Queueueueuueue position On Thu, 16 Dec 2004 15:18:10 +0100, E. Versaevel [EMAIL PROTECTED] wrote: When I call in (with an agent logged in) I get to hear the MOH on the client side, hover no matter how high the hold time is, I NEVER get an announcement over my queue position or my estimated wait time? Both the incoming call and the agent are on SIP channels. What is wrong ? Kind regards, E. Versaevel Would that be because this is the only call in queue? Try putting another call in queue and see what you get. -Chuji ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queueueueuueue position
Hello, I've got the following queue.conf: [testQ] music=jr_80 ;Bore the caller with some 80's music announce=queue-testQ;Announcement to play to the Agent answering strategy=ringall;Let all hell break lose timeout=60 ;We should answer within 60s retry=5 ; announce-frequenty=15 ;Tell them where the are every 15 seconds announce-holdtime=yes ; Give them an estimated hold time queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou ; (Thank you for your patience.) queue-lessthan = queue-less-than; (less than) join-empty=yes leavewhenempty=no member=Agent/1000 When I call in (with an agent logged in) I get to hear the MOH on the client side, hover no matter how high the hold time is, I NEVER get an announcement over my queue position or my estimated wait time? Both the incoming call and the agent are on SIP channels. What is wrong ? Kind regards, E. Versaevel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial Plan Problems
http://www.voip-info.org/wiki-Asterisk+config+extensions.conf+sorting Erik -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Ian Chilton Verzonden: dinsdag 14 december 2004 11:33 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] Dial Plan Problems Hi, I am having a few dial plan problems which I wondered if anyone would be able to help with. Firstly, I wanted to send 0800 calls through 1 sip provider and other 08xx calls through another. I have this: exten = _0800.,1,Dial(SIP/[EMAIL PROTECTED],30) exten = _0800.,2,Congestion exten = _08.,1,Dial(SIP/[EMAIL PROTECTED],30) exten = _08.,2,Congestion However, whichever way round I put these, 0800 calls still seem to go out of provider2. I fixed this as follows: exten = _08[1-9].,1,Dial(SIP/[EMAIL PROTECTED],30) exten = _08[1-9].,2,Congestion ..but, is there another way of getting this working without putting exceptions into other dial plan rules? - it works in this example but would be a pain in anything more complicated. For example, I had the same problem with a provider who has some service numbers in the format 09XX but the dial plan for these was not getting detected because of my _0. dial plan for national phone calls. Am I missing something here? Also, I can't seem to get the invalid extension working. I have a context which has some incoming calls sent to it. I then have an extension for each number. What I want is to play an invalid message if anyone comes into that context on another number. I tried this by commenting the entries for 1 of my working numbers out and putting this in: exten = i,1,Playback(invalid) exten = i,2,Hangup .. but when I call the number I commented out, Asterisk doesn't see the call (no logging) - I just get a beep and the call ends. If I put the extension for that number back in, it works fine. Any ideas? Thanks! --ian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] High(er) availability
Yes and no, what if Asterisk itself crashes? You would have a cluster full of machines, but no running server :) -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Kanwar Ranbir Sandhu Verzonden: dinsdag 7 december 2004 22:02 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] High(er) availability On Tue, 2004-07-12 at 10:54 +0100, E. Versaevel wrote: Hello, If one would like to build a redundant Asterisk setup, would it be possible to exchange the locationdb for the SIP users between then? I haven't tried this, but in my opinion it seems like it would give you what you're looking for: http://openssi.org/cgi-bin/view?page=openssi.html Actually, OpenSSI would work for just about anything, no? HTH, Ranbir -- Kanwar Ranbir Sandhu Systems Aligned Inc. www.systemsaligned.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] High(er) availability
Hello, If one would like to build a redundant Asterisk setup, would it be possible to exchange the locationdb for the SIP users between then? IE, the following setup: SIP Phones -- Asterisk SIP carrier | | --- Asterisk (standby) -- Asterisk is used as a PABX in this setup, so the sip phones register themselves at the asterisk machine and the asterisk machine calls out if necessary. What I would like to be able to do is if the first asterisk machine fails I want to have a 2nd machine standby. So the standby asterisk monitors the first asterisk and in case of a failure the standby asterisk takes over the IP of the 1st asterisk so the services continues (sync the conf file with rsync for example), however if the phones use a host=dynamic they wont be able to be called until they have reregistered themselves at the backup asterisk. Is there a SER like t_relay kinda thingy to let the backup know the locations of the Sip Phones? Kind regards, E. Versaevel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] High(er) availability
On Tue, 2004-12-07 at 10:54 +0100, E. Versaevel wrote: Hello, If one would like to build a redundant Asterisk setup, would it be possible to exchange the locationdb for the SIP users between then? Basically I would start with building redundancy in the the primary server, e.g. a ton of fans so one can break down without frying the box, redundant hot swappable power supplies, hot swappable disks in RAID1 or something like that. That will reduce the chance of the primary server going down due to hardware problems. That would be logical, also at least two ups (1 for each powersupply) Interesting problem as you need to be able to preserve state across multiple servers. Did you look at that realtime app that's part of CVS (maybe asterisk-addons)? If it stores the registration state of the phones in a DB then both servers should have no problem being aware of all regs if one of them fails. Haven't looked into that, I believe that's for realtime reading of the config files (which isn't realy an issue, just rsync em) Question is how do you make the backup server's Asterisk listen on the newly assigned IP of the primary server? Prolly a sip reload would solve that. Wouldn't SER be a better option? I'm thinking of giving the backup ser an alias the same as the primary, but it should not respond to ARP request (so no packets get there), in case of a failure it should start responding to ARP. SER isn't an option, I need a PABX, not a SIP Proxy. Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. Regards, Patrick Regards, E. Versavel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] High(er) availability
That would lead more to keepalived I think Would be an option, but I would have to use fixed IP addresses for the IP Phones (that should not be a problem) Erik E. Versaevel wrote: Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. VRRP, Virtual Redundancy Router Protocol, an option? Stefan de Konink ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] High(er) availability
Loosing calls wouldn't be to much of a problem I think, and it would be impossible to make a gracefull takeover if asterisk is in the mediastream. keepalived implements vrrp2 so that might be good enough. The problem lies in the registration data, but that could be solved by using fixed ip addresses for the phones. I need to setup a test environment, which I might just do :) Erik -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Tim Donahue Verzonden: dinsdag 7 december 2004 16:32 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] High(er) availability On Tue, 2004-12-07 at 15:47 +0100, Stefan de Konink wrote: E. Versaevel wrote: Which app do you use for monitoring the primary box and if it fails taking over the IP address by the backup one? I haven't found a suitable (active-active) app so far. Thinking of using heartbeat or something. VRRP, Virtual Redundancy Router Protocol, an option? Cisco claims that VRRP falls under one of their patents, so it could become an expensive option. There are several options out there at this point though that may be able to handle the needs for pre-empting the IP address. About 1 year ago the OpenBSD project wrote a patent-free alternative for VRRP called CARP. It allows for sharing of and automatic failover on an IP address. I have used it to build redundant firewalls that don't lose any state information when the connection drops. CARP is of course built into OpenBSD however I did find what looks to be a userland implementation for Linux. See www.ucarp.org for more information. There are other possible solutions as well, unfortunately I have not used any of these solutions they are just from brief google search. LVS (Linux Virtual Server) mentions VoIP services however I do not know if Asterisk would run in a cluster environment. There are also several sites that deal with high availibity from linux, the first one I noticed that looked like it had some really valuable information is www.linux-ha.org. Unfortunately this is all the easy part. The difficult part will be getting Asterisk to handle the failover gracefully. You probably don't want to lose all the SIP registration data and I have no idea if it will be possible to prevent you from losing the calls. You haven't named that as one of your goals, but it is always something to think about. -- Tim Donahue [EMAIL PROTECTED] Haynes Group, Incorporated ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial Plan Help
A DigitTimeout(3) will do wonders to (and fix the non existing priorities). Kind regards, E. Versaevel -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens [EMAIL PROTECTED] Verzonden: vrijdag 3 december 2004 21:52 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: [Asterisk-Users] Dial Plan Help All, I've got a problem here. We are using a Digium 4 T-1 board in our * server. The T-1's are ISDN. The problem I'm having is that we have an ivr setup so that when someone dials our DID it goes to the s extension and starts playing the ivr which is fine, but if someone dials an extension for example extension 200, it doesnt go to 200 it goes to extension 2. Seems like our server doesn't even wait for the rest of the digits dialed. soon as it sees 2 it goes straight to exten 2 and ignores the last two zeros therefore never reaching extension 200. any suggestions? i've enclosed a snipet below. TIA, -Jon exten=s,1,Answer exten=s,2,Wait(1) exten=s,3,Background(intro) exten=s,4,Background(ivrmenu) exten=i,1,Playback(invalid) exten=i,2,Goto(s|4) exten=200,Goto(office,102,1);forward to 102 in office context exten=201,Goto(office,110,1);forward to 110 in office context exten=1,1,Goto(office,102,1) exten=2,1,Goto(office,103,1) exten=3,1,Goto(office,104,1) exten=4,1,Goto(office,105,1) exten=5,1,Goto(office,106,1) exten=0,1,Goto(office,107,1) exten=t,1,Goto(office,108,1) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Call Monitor and soxmix error
Have you checked if nice
allso exists?
It tries to move the
soxmix to the background
Van:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Craig Waddington
Verzonden: woensdag 1 december
2004 15:56
Aan:
[EMAIL PROTECTED]
Onderwerp: [Asterisk-Users]
Asterisk Call Monitor and soxmix error
Asterisk Monitor seems to be working fine. Though the
problem I am having is the two files (in out) muxing.
I added ,m to the string, yet the call records two
files still, and I get the resulting error, at the bottom.
monitor executing ( nice -n 19 soxmix
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23-in.gsm
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23-out.gsm
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23.wav
rm -f
/var/spool/asterisk/monitor/rec_fr_1624672199_to_621950_at_01122004-13:48:23-*
)
nice: soxmix: No such file or directory
soxmix exists
exten = _8.,2,Monitor(gsm,${CALLFILENAME},m)
Path to soxmix = /usr/bin/soxmix
Asterisk seems to be looking in the wrong place for
it?
Is there a command line for soxmix to test muxing two
.gsm files ?
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RE: [Asterisk-Users] Can't hear playtones?
Before I start looking in the wrong places, does asterisk support the indications.conf on SIP channels? Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens E. Versaevel Verzonden: donderdag 25 november 2004 9:51 Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion' Onderwerp: [Asterisk-Users] Can't hear playtones? Hello, I would like the dialing party to know what happened to the call, since asterisk doesnt relay a sip error back to the originating sip channel (would be nice, a if (org_channel = sip dst_channel = sip, relay error to sip client) I want to set up audio feedback on the call status. Ive changed the county setting to NL in indications.conf and created this test extension: Exten = s,1, answer Exten = s, 2, playback(test) Exten = s, 3, playtones(busy) But I cant hear a busy tone on my sip phone, the call is answered, I hear the test file playback, but no busy tone. I tried to enter the values directly into playtones, but that didnt work either. Am I missing something? Kind regards, E. Versaevel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Problem!
If you want to Sip REGISTER your phone to asterisk change the
host=192.168.10.193 section of the [101] section to host=dynamic
Currently you are telling asterisk that sip user 101 is on host
192.168.0.193, which is you asterisk box, so when a call goes to 101,
asterisk sends it to itself and then tries to connect the incoming sip call
to 101, hence the loop :)
Kind regards,
E. Versaevel
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens Adnan Ahmed
Verzonden: maandag 22 november 2004 21:34
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] SIP Problem!
hi,
I am not registered my SIP Phone with Asterisk i spend almost one day
but find no luck.I know very well this is not kind a problem discussed
in this group but i try my best and all in vein so finally i am here
hoping you ppl helping me out.I discussed this problem in
asterisk's-users group and adding feedback from asterisk-users group my
configs are
sip.conf
[general]
port=5060
bindaddr=192.168.10.193
allow=all
[101]
username=101
type=friend
secret=12345678
host=192.168.10.193
context=from-sip
callerid=101101
defaultip=192.168.10.176
extensions.conf
[globals]
101=SIP/101
[incoming]
exten = s,1,Dial(Zap/1,20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${announce})
exten = s-NOANSWER,2,Goto(incoming,s,1)
exten = s,3,NoOp,$(CALLERID)
include = outgoing
include = from-sip
callerid=yes
[outgoing]
exten = _NXX,1,Dial/Zap/4/${EXTEN:0}
exten = _0N,1,Dial,Zap/4/${EXTEN:0}
exten = _0NX,1,Dial,Zap/4/${EXTEN:0}
exten = _0NXX,1,Dial,Zap/4/${EXTEN:0}
exten = 101,1,Dial(101,20)
include = from-sip
include = incoming
[sip]
exten = 101,1,Dial(${101,20})
exten = 101,2,VoicemailMain
exten = 101,3,Hangup
include = outgoing
include = from-sip
here are the console output : :-X ).
*cli --Starting simple switch on 'Zap/1-1'
Executing Dial( ,) in
new stack
Called 101
Got SIP Responce 482 Loop Detected back from 192.168.10.193
No one is available to answer qt this time
Executing VoiceMailMain( ,) in new stack
Playing 'vm-login' (language 'en' )
Username not entered
Executing Hangup( ,) in new stack
Spawn Extension (outgoing , 101, 3) exited non-zero on 'Zap/1-1'
Hangup 'Zap/1-1'
*clisip show registry
Host Username
Refresh State
*clisip show users
Username Secret Authen
Def.Context A/C
101 12345678md5,plaintext
sipNo
*clisip show peers
Name/UsernameHost Mask
Port Status
101/101192.168.10.195255.255.255.255
5060Unmonitored
*clisip show channels
PeerUser/ANRCall IDSeq
(Tx/Rx) LagJitterBuffer
0 active SIP channel(s)
Kindly pointout my mistakes/errors and helping me out.
Any Help Is Highly Appreciated.
Thanks in Advance.
Adnan Ahmed.
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[Asterisk-Users] Can't hear playtones?
Hello, I would like the dialing party to know what happened to the call, since asterisk doesnt relay a sip error back to the originating sip channel (would be nice, a if (org_channel = sip dst_channel = sip, relay error to sip client) I want to set up audio feedback on the call status. Ive changed the county setting to NL in indications.conf and created this test extension: Exten = s,1, answer Exten = s, 2, playback(test) Exten = s, 3, playtones(busy) But I cant hear a busy tone on my sip phone, the call is answered, I hear the test file playback, but no busy tone. I tried to enter the values directly into playtones, but that didnt work either. Am I missing something? Kind regards, E. Versaevel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk not relaying SIP messgaes
Hello, Im using asterisk to relay sip to another SIP provider, Ive setup a friend in sip.conf for my softphone and a user peer section for the SIP provider, when my softphone calls out to the sip provider and the sip provider returns an error (404 Not Found for example) the sip message is not relayed back to my sip phone, it just sits and waits for a timeout. Is it possible to relay the returned SIP (error) message to the softphone? Kind reagards, E. Versaevel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Commercial g723.1 license for asterisk
Which is not for commercial use Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Vinicius Viana Verzonden: dinsdag 23 november 2004 14:57 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: RES: [Asterisk-Users] Commercial g723.1 license for asterisk I never used, but there are one open beta codec g723.1 from the makers of the open g729 codec that uses Intel IPP library. http://www.readytechnology.co.uk/open/g723.1/ Regards, Vinicius -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]Em nome de kido noagbodji Enviada em: terça-feira, 23 de novembro de 2004 09:34 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [Asterisk-Users] Commercial g723.1 license for asterisk Hi all, Is there any commercial g723 license for asterisk? Where can it be purchased? Has somebody used it? Thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange Fromuser behavior?
Strange things are happening at my asterisk box :) I've got asterisk setup to dail out with sip to my SIP provider. I've got NO fromuser/fromdomain stuff setup in my sip.conf When I place a call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads [EMAIL PROTECTED] instead of [EMAIL PROTECTED] Any idea how this is possible? Kind reagards, E. Versaevel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Strange Fromuser behavior?
Hmm, a bit closer, Asterisk seems to do the asterisk@ part only with non numeric usernames, ie [EMAIL PROTECTED] stays [EMAIL PROTECTED] but [EMAIL PROTECTED] turns into [EMAIL PROTECTED] -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens E. Versaevel Verzonden: maandag 22 november 2004 13:27 Aan: 'Asterisk Users Mailing List - Non-Commercial Discussion' Onderwerp: [Asterisk-Users] Strange Fromuser behavior? Strange things are happening at my asterisk box :) I've got asterisk setup to dail out with sip to my SIP provider. I've got NO fromuser/fromdomain stuff setup in my sip.conf When I place a call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads [EMAIL PROTECTED] instead of [EMAIL PROTECTED] Any idea how this is possible? Kind reagards, E. Versaevel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Setup/SIP routing
I'm still stuck this/my problem.
Even if I create a friend entry and register my softphone directly to
Asterisk, the Dail(${EXTEN},entity) seems to replace the From: part with the
From: 349525 sip:[EMAIL PROTECTED]:5065 part instead of the
From: 349525 sip:[EMAIL PROTECTED]
So if I would add that incoming call to my addressbook the sip URI is wrong.
I'm thinking something in the way of fromuser=${SIPCALLID} would be needed
for this?
I'm also not able to get Asterisk out of the mediastream, I've set the
canreinvite options to yes, but still asterisk stays in the stream.
I've made a SIP scenario trace of the callsetup, I'm a bit puzzled by the 2
time call setup?
Kind regards,
E. Versaevel
-Oorspronkelijk bericht-
Van: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Namens E. Versaevel
Verzonden: donderdag 18 november 2004 13:53
Aan: [EMAIL PROTECTED]
Onderwerp: [Asterisk-Users] RE: Setup/SIP routing
The problem is that that should be dynamic :/
Take a look at this sip msg:
INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0
Max-Forwards: 10
Record-Route: sip:[EMAIL PROTECTED];ftag=as3f718642;lr=on
Via: SIP/2.0/UDP ser.box;branch=z9hG4bK93dc.71ad80b3.0
Via: SIP/2.0/UDP ser.box:5065;branch=z9hG4bK513b584d
From: 349525 sip:[EMAIL PROTECTED]:5065;tag=as3f718642
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]:5065
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Thu, 18 Nov 2004 12:46:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 220
P-hint: USRLOC
v=0
o=root 26383 26383 IN IP4 ser.box
s=session
c=IN IP4 ser.box
t=0 0
m=audio 14682 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
As you can see the from user is not correct, this should be
[EMAIL PROTECTED] If a user adds this entry to a phonebook, the contact
info will be wrong.
-Oorspronkelijk bericht-
Van: Benjamin on Asterisk Mailing Lists
[mailto:[EMAIL PROTECTED]
Verzonden: donderdag 18 november 2004 11:41
Aan: E. Versaevel
Onderwerp: Re: [Asterisk-Users] Setup/SIP routing
Hi
On Thu, 18 Nov 2004 11:32:08 +0100, E. Versaevel [EMAIL PROTECTED] wrote:
However, I'm having troubles routing incoming sip traffic to SER,
asterisks
keeps messing up the form header (replacing it by the dialed context, ie
[EMAIL PROTECTED] )
You can control what Asterisk puts into the FROM header through the
parameters fromuser and fromdomain in sip.conf.
regards
benjamin
--
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.
NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
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IpTel Asterisk
SER
SoftPhone
| |
| |
CallPFrameTime
| |
| |
|F1 INVITE (sdp)-|
| | 1
PF:70 09:27:15.
| |
| |
|-- Trying 100 F2|
| | 1
PF:71 09:27:15.7783
| |
| |
| |F3 INVITE
(sdp)-|
| 2 PF:72 09:27:15.7789
| |
| |
| |-- trying -- your call is
important to us 100 F4| | 2
PF:73 09:27:15.7795
[Asterisk-Users] Setup/SIP routing
Hello, I'm still kinda new to asterisk, but I'm trying to setup the following situation: Aterisk running at port 5065, SER running at 5060 (done that, works fine) Load of SIP clients registering at SER (no problem), SER routing the SIP traffic to Asterisk (no problem). Incomming: *) Asterisk registers at a SIP provider which sends SIP traffic to my host, but not addressed to the registered user. *) Asterisk has to pickup that sip traffic and relay it to ser at the same host, but should not mess up the addressed user. *) Ser performs routing (no problem here) Outgoing: *) SER sends outbound traffic to Asterisk (no problem) *) Asterisk has to dial out to that number, but is challenged by my SIP provider, asterisk answers challenge (should work by setting up a peer) The reason for this setup is that my SIP Provider requires me to register at his proxy (SER can't do that) and has to authenticate outbound calls (SER can't do that either) However, I'm having troubles routing incoming sip traffic to SER, asterisks keeps messing up the form header (replacing it by the dialed context, ie [EMAIL PROTECTED] ) Any ideas if this setup is even possible? Kind regards, E. Versaevel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Setup/SIP routing
The problem is that that should be dynamic :/ Take a look at this sip msg: INVITE sip:[EMAIL PROTECTED]:5061 SIP/2.0 Max-Forwards: 10 Record-Route: sip:[EMAIL PROTECTED];ftag=as3f718642;lr=on Via: SIP/2.0/UDP ser.box;branch=z9hG4bK93dc.71ad80b3.0 Via: SIP/2.0/UDP ser.box:5065;branch=z9hG4bK513b584d From: 349525 sip:[EMAIL PROTECTED]:5065;tag=as3f718642 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5065 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Thu, 18 Nov 2004 12:46:51 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 220 P-hint: USRLOC v=0 o=root 26383 26383 IN IP4 ser.box s=session c=IN IP4 ser.box t=0 0 m=audio 14682 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - As you can see the from user is not correct, this should be [EMAIL PROTECTED] If a user adds this entry to a phonebook, the contact info will be wrong. -Oorspronkelijk bericht- Van: Benjamin on Asterisk Mailing Lists [mailto:[EMAIL PROTECTED] Verzonden: donderdag 18 november 2004 11:41 Aan: E. Versaevel Onderwerp: Re: [Asterisk-Users] Setup/SIP routing Hi On Thu, 18 Nov 2004 11:32:08 +0100, E. Versaevel [EMAIL PROTECTED] wrote: However, I'm having troubles routing incoming sip traffic to SER, asterisks keeps messing up the form header (replacing it by the dialed context, ie [EMAIL PROTECTED] ) You can control what Asterisk puts into the FROM header through the parameters fromuser and fromdomain in sip.conf. regards benjamin -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip relay with asterisk
I've got the following setup: SIP Client -- SER -- Asterisk -- Iptel.org SIP account I'm now trying to place an outgoing call, which has to be authenticated at the iptel.org proxy server (which ser can't do) but I seem to be getting 407 packets with the IP of the asterisk machine as realm. SIP ClientSER * Iptel.org Invite --- Invite - Invite --- --- 407 realm ip_of_* Afaik the realm iptel.org should provide to * should be iptel.org instead of the IP of the * box (which would indicate an error at iptel?) What I'm trying to archive is that the * box authenticated the calls to iptel and then leave the call alone (so I will have to find out how to get * out of the media path) I'm still new to all this, but I think this could work. Kind regards, E. Versaevel Extensions.conf [sip_in_from_carrier] exten = _XX, 1, Dial(SIP/[EMAIL PROTECTED],20,r) ;Not a 10 digit number exten = s,1,Answer exten = s,2,MusicOnHold() exten = s,3,Hangup ;Timeout exten = t,1,Answer exten = t,2,Background(conf-invalid) ;exten = t,3,MusicOnHold() exten = t,4,Hangup ;Hangup exten = h,1,Hangup [sip_in_from_ser] exten = _., 1, Dial(SIP/[EMAIL PROTECTED],20,r) ;Not a 10 digit number exten = s,1,Answer exten = s,2,MusicOnHold() exten = s,3,Hangup ;Timeout exten = t,1,Answer exten = t,2,Background(pin-invalid) ;exten = t,3,MusicOnHold() exten = t,4,Hangup ;Hangup exten = h,1,Hangup [default] exten = s, 1, Background(conf-invalid) exten = s, 2, Hangup Sip.conf [general] port=5065 disallow=all allow=ulaw register = asterisk:[EMAIL PROTECTED] ;Incomming from ser register = iptel:[EMAIL PROTECTED]/iptel_alias ;Incomming from iptel [sip.carrier] type=user realm=iptel.org username=iptel secret=iptel host=sip.iptel.org canreinvite=no context=sip_in_from_carrier [sip.carrier] type=peer host=sip.iptel.org context=sip_in_from_carrier [sip.ser] type=user realm=sermachine host=sermachine canreinvite=no context=sip_in_from_ser [sip.ser] type=peer host=sermachine context=sip_in_from_ser ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP clients -- SE R -- Asterisk -- carrier/gateway
Hello,
I'm currently trying to setup a SIP environment for VoIP calling for my
final school project, so I'm just working with VoIP/SIP for 2 weeks.
I'm using SER as a SIP proxy server, but the carrier/gateway I am using for
calling to/from PSTN is requiring me to register at their server and
authorize outgoing calls, which is something SER won't do. So I got the idea
to use asterisk between the PSTN carrier and SER for the authorization,
since Asterisk can register and auth itself.
SIP - SIP -SIP- PSTN
-| |---| |---| |--
- - -
SER Asterisk Carrier
-- auth stuff --
-- sip relay --
So all asterisk needs to do is register itself at the carrier (I've got that
to work with sip.conf) and relay the incoming calls to ser for further
routing, I got that to work a bit.
I've setup 2 sip extensions in extensions.conf in the default context
(asterisk itself is listening at 5065)
[globals]
SERADDRESS=myserbox:5060
CARRIER=carrier:5060
[default]
exten = sip_incomming_from_carrier, 1,
Dial(SIP/[EMAIL PROTECTED],20,r)
exten = sip_incomming_from_ser, 1,
Dial(SIP/[EMAIL PROTECTED],20,r)
When I get an incoming call from the carrier it gets routed to the SER
server (wrong SIP uri however, it now gets
[EMAIL PROTECTED], but that's due to the {EXTEN}, that
should be [EMAIL PROTECTED]), but replies from my SER box are not
getting back to the carrier, so if a user is not found (SIP/2.0 404) I see
that message on the Asterisk console (Got SIP response 404 Not Found back
from myserbox), but it isn't relayed to the carrier.
I'm also talking with the carrier about skipping the authorization (or
moving it to a lower layer IE vpn oid), but I like to have a solution ready
if the carrier doesn't want that.
Kind regards,
E. Versaevel
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