[Asterisk-Users] Automatically send monitored call files by e-mail
Hi, I want to automatically send the sound files generated by asterisks monitor functions to a certain email address. My knowledge of shell scripting leaves a lot to desire, so I was hoping maybe on of you guys already did this and might provide me with an example of what to do :) Best regards, Anders ___ 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] Re: Zap channel calling back after hangup (due to polarity CID detection)
That issue is fixed in the CVS HEAD version of asterisk. There are a couple of workarounds possible with 1.0.6. Check the bugtracker for the bug where it was implemented for more information. (sorry, don't remember the bug-number and don't have time to look it up right now). You might also try the following patch: http://evil.gnarf.org/creativity/asterisk/20050110-answeronpol arity.diff which is what is in HEAD. But I don't know if it will work against asterisk stable. If you know C it should be no problem adapting it though. I'm running CVS-HEAD-02/28/05-23:18:39 at the moment, and it still happens. I've seen the bugs in Mantis, but the answeronpolarity doesn't seem to make any difference ... /Anders ___ 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] Re: Zap channel calling back after hangup (dueto polarity CID detection)
That issue is fixed in the CVS HEAD version of asterisk. There are a couple of workarounds possible with 1.0.6. Check the bugtracker for the bug where it was implemented for more information. (sorry, don't remember the bug-number and don't have time to look it up right now). I'm running CVS-HEAD-02/28/05-23:18:39 at the moment, and it still happens. I've seen the bugs in Mantis, but the answeronpolarity doesn't seem to make any difference ... Could you post a debug-log of when it happens? (enable debug and verbose in logger.conf) This is the log: Mar 1 13:57:06 VERBOSE[15245]: -- Accepting AUTHENTICATED call from 213.136.48.247: requested format = alaw, requested prefs = (), actual format = alaw, host prefs = (alaw|ulaw|gsm), priority = mine Mar 1 13:57:06 VERBOSE[15253]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mIAX2/[EMAIL PROTECTED];37;40m, [1;35;40mZap/g2/0850007225|120|W[0;37;40m) in new stack Mar 1 13:57:06 DEBUG[15253]: Dialing '0850007225' Mar 1 13:57:06 DEBUG[15253]: Deferring dialing... Mar 1 13:57:06 VERBOSE[15253]: -- Called g2/0850007225 Mar 1 13:57:06 DEBUG[15245]: Ooh, voice format changed to 8 Mar 1 13:57:06 DEBUG[15253]: Exception on 19, channel 4 Mar 1 13:57:06 DEBUG[15253]: Got event Hook Transition Complete(12) on channel 4 (index 0) Mar 1 13:57:08 DEBUG[15253]: Exception on 19, channel 4 Mar 1 13:57:08 DEBUG[15253]: Got event Dial Complete(9) on channel 4 (index 0) Mar 1 13:57:08 DEBUG[15253]: Enabled echo cancellation on channel 4 Mar 1 13:57:08 DEBUG[15253]: Engaged echo training on channel 4 Mar 1 13:57:10 DEBUG[15253]: Exception on 19, channel 4 Mar 1 13:57:10 DEBUG[15253]: Got event Dial Complete(9) on channel 4 (index 0) Mar 1 13:57:10 DEBUG[15253]: Echo cancellation already on Mar 1 13:57:10 DEBUG[15253]: Dropping duplicate answer! Mar 1 13:57:10 VERBOSE[15253]: -- Zap/4-1 answered IAX2/[EMAIL PROTECTED] Mar 1 13:57:10 DEBUG[15253]: Exception on 19, channel 4 Mar 1 13:57:10 DEBUG[15253]: Got event Polarity Reversal(17) on channel 4 (index 0) Mar 1 13:57:10 DEBUG[15253]: Ignore switch to REVERSED Polarity on channel 4, state 6 Mar 1 13:57:14 DEBUG[15245]: Raw Hangup 69.73.19.178:4569, src=3, dst=143 Mar 1 13:57:15 VERBOSE[15239]: -- Remote UNIX connection Mar 1 13:57:20 DEBUG[15245]: Immediately destroying 2, having received hangup Mar 1 13:57:20 DEBUG[15253]: Didn't get a frame from channel: IAX2/[EMAIL PROTECTED] Mar 1 13:57:20 DEBUG[15253]: Bridge stops bridging channels IAX2/[EMAIL PROTECTED] and Zap/4-1 Mar 1 13:57:20 DEBUG[15253]: Hangup: channel: 4 index = 0, normal = 19, callwait = -1, thirdcall = -1 Mar 1 13:57:20 DEBUG[15253]: disabled echo cancellation on channel 4 Mar 1 13:57:20 DEBUG[15253]: Set option TDD MODE, value: OFF(0) on Zap/4-1 Mar 1 13:57:20 DEBUG[15253]: Updated conferencing on 4, with 0 conference users Mar 1 13:57:20 VERBOSE[15253]: -- Hungup 'Zap/4-1' Mar 1 13:57:20 DEBUG[15253]: Exiting with DIALSTATUS=ANSWER. Mar 1 13:57:20 VERBOSE[15253]: == Spawn extension (norby, 90850007225, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]' Mar 1 13:57:20 VERBOSE[15253]: -- Executing [1;36;40mHangup[0;37;40m([1;35;40mIAX2/[EMAIL PROTECTED];37;40m, [1;35;40m[0;37;40m) in new stack Mar 1 13:57:20 VERBOSE[15253]: == Spawn extension (norby, h, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]' Mar 1 13:57:20 DEBUG[15253]: We're hanging up IAX2/[EMAIL PROTECTED] now... Mar 1 13:57:20 DEBUG[15253]: Really destroying IAX2/[EMAIL PROTECTED] now... Mar 1 13:57:20 VERBOSE[15253]: -- Hungup 'IAX2/[EMAIL PROTECTED]' Mar 1 13:57:22 VERBOSE[15246]: == Starting post polarity CID detection on channel 4 Mar 1 13:57:22 VERBOSE[15256]: -- Starting simple switch on 'Zap/4-1' Mar 1 13:57:22 DEBUG[15256]: Receiving DTMF cid on channel Zap/4-1 Mar 1 13:57:23 DEBUG[15256]: Exception on 19, channel 4 Mar 1 13:57:23 DEBUG[15256]: Got event Ring/Answered(2) on channel 4 (index 0) Mar 1 13:57:23 DEBUG[15256]: Setting IDLE polarity due to ring. Old polarity was 1 Mar 1 13:57:23 DEBUG[15256]: CID got string '' Mar 1 13:57:23 DEBUG[15256]: No cid detected Mar 1 13:57:23 DEBUG[15256]: CID is '', flags 8 Mar 1 13:57:23 VERBOSE[15256]: -- Executing [1;36;40mSetCallerID[0;37;40m([1;35;40mZap/4-1[0;37;40m, [1;35;40m0[0;37;40m) in new stack Mar 1 13:57:23 VERBOSE[15256]: -- Executing [1;36;40mDial[0;37;40m([1;35;40mZap/4-1[0;37;40m, [1;35;40mSIP/400Zap/1|45|tw[0;37;40m) in new stack Mar 1 13:57:23 DEBUG[15256]: Setting NAT on RTP to 0 Mar 1 13:57:23 DEBUG[15256]: Outgoing Call for 400 Mar 1 13:57:23 VERBOSE[15256]: -- Called 400 Mar 1 13:57:23 VERBOSE[15256]: -- Called 1 Mar 1 13:57:23 VERBOSE[15256]: -- Zap/1-1 is ringing Mar 1 13:57:23 DEBUG[15256]: Requested indication 3 on channel Zap/4-1 Mar 1 13:57:23 DEBUG[15244]: (Provisional) Stopping retransmission (but retaining packet) on '[EMAIL
RE: [Asterisk-Users] Re: Zap channel calling back after hangup (duetopolarity CID detection)
The problem is the Got event Ring/Answered(2) line. Normally, a ring should not be detected and the DTMF-cid times out and no incoming call is registered. Make sure you load wctdm with the parameter 'opermode=SWEDEN', it might help. You might also try to increase the 'RING_DEBOUNCE' define in wctdm.c. If you load wctdm with debug=1 it prints some useful information to the kernel log for tuning the RING_DEBOUCE. opermode=SWEDEN did unfortunately not help. I increased the RING_DEBOUNCE and recompiled zaptel, and now it doesn't call back. However it seems it tries, but times out now. --- Starting post polarity CID detection on channel 4 -- Starting simple switch on 'Zap/4-1' Mar 1 17:48:11 WARNING[23918]: chan_zap.c:5351 ss_thread: DTMFCID timed out waiting for ring. Exiting simple switch --- /Anders ___ 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] Re: Zap channel calling back after hangup (duetopolarity CID detection)
The problem is the Got event Ring/Answered(2) line. Normally, a ring should not be detected and the DTMF-cid times out and no incoming call is registered. Make sure you load wctdm with the parameter 'opermode=SWEDEN', it might help. You might also try to increase the 'RING_DEBOUNCE' define in wctdm.c. If you load wctdm with debug=1 it prints some useful information to the kernel log for tuning the RING_DEBOUCE. Ok, I tried opermode=SWEDEN. That did not make any difference. After increasing the RING_DEBOUNCE from 64 to 128 ms it no longer calls me back, but it tries: ___ 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] Re: Zap channel calling back after hangup(duetopolarity CID detection)
That is the expected behaviour. =) It is so because when YOU hang up, asterisk hangs up the channel and destroys it. A moment later (actually up to 90sec in sweden) the pstn disconnects the call and signals a polarity reversal. That reversal causes the above. I have a solution but I don't know enough about chan_zap to do it, and don't have the time to learn. The solution is basically to wait(with a timeout) for a reversal indicating that the PSTN has terminated the call before destroying the channel. But I don't really know where to put it. Ah, thanks :) At least it works now - I can live with the WARNING message. Hopefully someone with a better knowledge of programming than I have will solve this some day ... Best regards, Anders ___ 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] Zap channel calling back after hangup (due to polarity CID detection)
Today I received a TDM11B (1 FXO and 1 FXS) and got it installed just fine. I bought the card mainly to get caller ID to work properly in Sweden, and that works just fine. However, if the called or calling party hangs up after I hangup my SIP channel, polarity CID detection kicks in and dials a couple of signals to my incoming context. This happens with Asterisk 1.0.6 and CVS-HEAD. I have tried various combinations of hanguponpolarityswitch and answeronpolarityswitch (and without them). I'm sure this has been solved several times, but I can't seem to find it anywhere - any help appreciated :) My zapata.conf at the moment looks as follows: ;* ; zapata.conf ;* ; [channels] ; ; TDM11B plugged into PSTN ; signalling=fxo_ks busydetect=yes echocancel=yes echocancelwhenbridged=yes echotraining=400 usecallerid=yes callerid=Siemens 450 group=1 threewaycalling=yes transfer=yes context=norby channel = 1 signalling=fxs_ks txgain=0.0 rxgain=15.0 usecallerid=yes callerid=asreceived cidsignalling=dtmf cidstart=polarity transfer=no group=2 context=analog_in answeronpolarityswitch=no hanguponpolarityswitch=yes channel = 4 ___ 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] IAX2 Softphone
The only issue I have had with it so far, and it may be a misconfiguration problem as I am certainly a newb, is that when I dial a number that sends it over to my POTS line, I get ringing from the softphone and the POTS line. When to POTS line answers, the softphone continues to ring. Nice work indeed. I am running it on a laptop under Windows XP w/ SP2 and it is working well. As I stated before the only issue I have had so far is the ringing issue. The ringing issue seems to be a problem with the iaxlib - I have the same problem with both this program and Diax. I can partly be solved by adding an Answer() as the first priority in outgoing contexts. After doing that I instead get problems when the called part is busy - I get no audio notification of the busy status (either in Mediax or Diax. /Anders ___ 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] IAX2 Softphone
With an X100P card I get the PSTN line ringtone and/or busy tone in DIAX when an outgoing call is in progress. No need to have an Answer before... I forgot to mention that I'm connecting to the PSTN with a SIP connection to my provider. /Anders ___ 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] choppy sound after 15 minutes in a call
I'm using X-Pro connected to an asterisk server (CVS-HEAD-01/27/05-23:17:07) and after about 15 minutes in a call I get a lot of noise in my end. I don't think the other part of the call hears it. After some 10 seconds or so everything is fine again. In my CLI I get NOTICE[32322]: RTP Transmission error to 85.xxx.xxx.xxx:35162: Operation not permitted. I get it on calls to the PSTN through my X100P (clone) as well as call connected through my IP telephony provider. I have also tried SJ-Phone and it happens with that as well. At the moment my asterisk server is on a public IP adress, and my client connects to it through an Intertex IX66 router. Before getting the router, I had dual NICs in the linux box and connected the client directly with the same problems... I've searched the lists but haven't been able to find any good answer, so any help is greatly appreciated :) /Anders ___ 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] Bootable Asterisk CD ?
Check out http://www.voip-info.org/wiki-Asterisk+Bootable+CDROM. /Anders A while ago, I saw some threads on booting linux w/ asterisk from a CF card. I have also seen CD installs of Asterisk, which require a hdd. Has anyone come up with a bootable cd (like a Live CD), that creates a ramdisk and runs asterisk, without touching the hard disk ? It would be a good tool to demo asterisk, without actuall installing linux. I looked at AstWind, but I dont think you can use the Console Channel driver with that. ___ 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] Broadvoice / * re-register issues
I think you might have to add the line below to [sip.broadvoice.com]: insecure=very I know that it's required for other services, and probably with broadvoice as well. /Anders Ok, so I have the following SIP.CONF: [general] context=default port=5060 bindaddr=10.1.1.200 externip = XX.XXX.XX.XX localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register = [EMAIL PROTECTED]:X:[EMAIL PROTECTED] ice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=## context=default dtmfmode=inband canreinvite=no disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw ___ 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 = Silence ???
If you use RH or Fedora the included mpg123 doesn't work. Check http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20redhat for info on how to fix it. /Anders Having played with the music on hold settings for 2 days now trying to get it to work I am stumped and need help from the group. Have a standard * installation with FXO cards and two working SIP phones (x-lite). I am able to make TDM-SIP, SIP-TDM, and SIP-SIP calls with no problem. When ever I place a call on hold (or any other time MOH should be played) I hear nothing but silence. Have checked settings in musiconhold.conf and read wiki on subject to no avail. Have implemented a test number in extensions.conf as follows... exten = 6601,1,Answer exten = 6601,2,WaitMusicOnHold(30) Console displays the following... -- Executing Answer(SIP/mmurdock-81b4, ) in new stack -- Executing WaitMusicOnHold(SIP/mmurdock-81b4, 30) in new stack -- Started music on hold, class 'default', on SIP/mmurdock-81b4 -- Stopped music on hold on SIP/mmurdock-81b4 Hear nothing but silence... musiconhold.conf has following... [classes] default = mp3:/var/lib/asterisk/mohmp3 Where do I look next? -- Mike ___ 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] Softphone with subscribe/notify support
Are there any good SIP softphones with support for SUBSCRIBE/NOTIFY, like the SNOM phones have? I tried one from Estara, but it costs $100 and it crashed when I tried to run it (maybe I could have make it run, but any software that crashes after running install/config wizard is not something I'll buy for $100). /Anders ___ 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] IAXTEL Configuration
Hi, I think you should remove the [iaxtel_out] from iax.conf This is a snip from mine iax.conf: [general] register = user:[EMAIL PROTECTED] [iaxtel] type=user context=incoming auth=rsa inkeys=iaxtel You then can modify extensions.conf to handle outgoing calls. See http://www.iaxtel.com/setup.html (which is where I got my settings). /Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Robins Sent: den 21 december 2004 22:52 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAXTEL Configuration I signed up for an IAXTEL account and have been trying, unsuccessfully, to get it working. In IAX.CONF I have: [iaxtel_out] type=peer host=iaxtel.com username=USERNAME secret=SECRET auth=rsa inkeys=iaxtel [iaxtel] type=friend context=incoming host=iaxtel.com auth=rsa inkeys=iaxtel However, when I start Asterisk, I get the following warning: [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found Dec 21 15:44:04 WARNING[24873]: chan_iax2.c:6602 build_user: Cannot allow unknown format 'iaxtel.com' Dec 21 15:44:04 WARNING[24873]: chan_iax2.c:6497 build_peer: Cannot allow unknown format 'iaxtel.com' For some reason, it does not like the host= lines. I've replaced 'iaxtel.com' with their IP, but that gives the same error. Please assist. Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ 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] Setting up asterisk for one user in private ipNAT.
Hmm... For outgoing connections (from the softphone) that's true, but if you want asterisk to send a call to the softphone the default would be to send it to port 5060 (which is already taken by asterisk). If the softphone is setup to register to asterisk on another port, there should be no problem. At least this is how I thought it worked ... /Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Hill Sent: den 19 december 2004 00:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Setting up asterisk for one user in private ipNAT. that isn't quite how ports work.. True, asterisk listens for udp connections on 5060. But the softphone won't make its outgoing connection on 5060. The OS will automatically choose an unused port number for the outgoing connection. So (for example) you might have the softphone talking on port 23107 to asterisk on 5060, and asterisk on 5060 talking back to the softphone on 23107. No port conflict. Now one place where you could have conflict is if asterisk is trying to use your soundcard for its console. Then the softphone client may have trouble getting the soundcard port opened. Greg ___ 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] Setting up asterisk for one user in private ip NAT.
I've never tried softphones on Linux, but my guess is that since you run
kphone and asterisk on the same server you get a port conflict. If the
client uses port 5060 (default sip port) it would defenitely have problem
connecting to an asterisk on the same port.
Maybe you can change the kphone settings to use some other port or something
:)
/Anders
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Alex Polite
Sent: den 18 december 2004 14:53
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Setting up asterisk for one user in
private ip NAT.
Hi.
I've just bought SIP telephony service from a Swedish telco.
I've managed to make and receive calls with kphone.
Now I want to set up asterisk to be able to add fancy
features like voice mail and recording conversations. But
first I have to get the basic setup right. I'm running
asterisk and kphone on the same machine, behind at NAT-router.
When I make a call (from my regular phone) to the SIP-number
I get a busy signal and I see my regular phone number in the
debug output of Asterisk. I guess that means I'm doing
something right.
The problem now is that I can't get kphone or linphone to
connect to asterisk. Trying to connect from kphone to
asterisk does not generate any messages in the asterisk debug
output. Non what so ever.
Which has me thinking that ip might be something with the
hostnames/ip-addresses that's not right?
What does bindaddr do? I've tried changing it to my private
IP but that doesn't make any difference.
I know that I'm not being very specific in my questions but I
feel that I need some handholding here. Some tests that I can
run, for example, to find out if my Asterisk setup is kosher.
So, will someone please hold my hand in this scary land of VOIP?
Alex
Here are my config files so far.
sip.conf
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
externip = public ip of router
localnet = 192.168.0.0/255.255.255.0 ; Internal
NETWORK address
allow=ulaw
allow=alaw
allow=gsm
allow=all
nat=yes
register = :[EMAIL PROTECTED]/1000
[alex]
type=friend
host=dynamic
username=alex
secret=
context=outgoing
[rix]
type=peer
username=xx
fromuser=xx
secret=
host=astrofix.rixtele.com
fromdomain=astrofix.rixtele.com
context=sip-in
insecure=very
nat=yes
extensions.conf
[default]
exten = 1000,1,Dial(SIP/alex||t)
[sip-in]
exten = 1000,1,Dial(SIP/alex||t)
[outgoing]
exten = _0.,1,Dial(SIP/rix/${EXTEN}|20|t)
.qt/kphonerc
[Registration]
AutoRegister=No
SipServer=
SipUri=Alex Polite sip:[EMAIL PROTECTED] UserName=alex qValue=
--
Alex Polite
http://polite.se
___
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] Incoming call problem
There is no way for asterisk to know which extension you want to call when you have a single analog line. You can either send the call to an extension or group, or you could create a menu system that allow the caller to select which extension to call (press 1 for Steve, 2 for Dave etc.). /Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of R A Sent: den 15 december 2004 21:20 To: Ning Zhou; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Incoming call problem hi all I have a x100p install in my server to conect my sip extensions whith a PABX. When i call from a sip phone to a pstn number it work well, but to call from pstn line to asterisk only can do it to especific phone doing this: exten = s,1,Dial(SIP/116) the problem is that i want to call any of my sip extensions not one directly. thanks in advance wert __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ 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] Dialing out to 2 clients simultaneously
Hi, I don't think any SIP server would allow you to register more than once with the same login information. What you can do in asterisk is setup two different entries in sip.conf and then use extensions.conf to dial both. Example from extensions.conf [default] exten = 1000,1,Dial(SIP/user1SIP/user2,60,t) exten = 1000,2,Congestion exten = 1000,3,Hangup exten = 1000,102,Busy /Anders -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: den 13 december 2004 14:39 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Dialing out to 2 clients simultaneously Hmmm that's bad... This is the last issue I have which makes that I can't get rid of the SER proxy in front of asterisk.. Want to get rid of it Are there any plans to change this design?? (that multiple UA's can register to one peer?) Niels ___ 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
