Re: [Asterisk-Users] OT?: International number parsing
There is second single digit code - 7 (Russia). On Sat, 2006-01-28 at 18:41 +0100, Francesco Peeters (Asterisk) wrote: Only one country has a single digit code: USA = 1 ___ --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] spandsp Error
Line 103 in Makefile has multiple spaces at the beginning instead of TAB character.
On Mon, 2006-01-23 at 10:19 +0800, Ronald Wiplinger wrote:
I cannot see it
make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk/pbx'
/bin/sh: curl-config: command not found
make[1]: Entering directory `/usr/local/src/svn-versions/asterisk/apps'
Makefile:103: *** missing separator. Stop.
make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk/apps'
make: *** [depend] Error 1
Makefile:
93install: all
94for x in $(APPS); do $(INSTALL) -m 755 $$x
$(DESTDIR)$(MODULES_DIR) ; done
95rm -f $(DESTDIR)$(MODULES_DIR)/app_cut.so
96rm -f $(DESTDIR)$(MODULES_DIR)/app_datetime.so
97rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so
98
99 app_curl.so: app_curl.o
100$(CC) $(SOLINK) -o $@ ${CYGSOLINK} $ ${CYGSOLIB} $(CURLLIBS)
101
102 app_rxfax.so : app_rxfax.o
103 $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
104
105 app_txfax.so : app_txfax.o
106 $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff
107
108 app_sql_postgres.o: app_sql_postgres.c
109$(CC) -pipe -I$(CROSS_COMPILE_TARGET)/usr/local/pgsql/include
-I$(CROSS_COMPILE_TARGET)/usr/include/postgresql $(CFLAGS) -c -o
app_sql_postgres.o app_sql_postgres.c
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Re: [Asterisk-Users] Alphanumeric pattern match in extensions.conf
The match doesn't work because n in conf will never match to the
letter n (it's a pattern for a digit).
try _co[n]f. instead.
On Fri, 2006-01-06 at 10:33 -0800, Dan Austin wrote:
I need to match an incoming call based on a prefixed string, and this
solution was suggested to me some time back.
exten = _conf.,1,Answer
exten = _conf.,2,MeetMe(${EXTEN:4}|d)
exten = _conf.,3,Hangup
However incoming calls never match this pattern, and I cannot
find any evidence in the wiki or on google that such a pattern
is valid. I'm currently running a SVN trunk, but have tested
with 1.0.X and 1.2.X.
Is anyone using alphanumeric patterns in their dialplan?
Dan
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Re: [Asterisk-Users] Via Epia
I run * on VIA EPIA M1 on gentoo. Here is my ~/.asterisk.makeopts: K6OPT = -DK6OPT DEBUG= ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run/asterisk PROC=i686 On Sat, 2005-12-10 at 12:36 +0100, Maciej Kietlinski wrote: Does anyone has some experience in installing * on Via Epia. I am struggling with it for about two days. And when I finally managed to install asterisk 1.0.9 after starting it I get this error or whatever: - Illegal instruction I changed the variable in makefile to i586 (I also tried i686 because that is what my uname - m says) but still I get the same problem. I use Debian with 2.4.30 kernel. Does anyone has some experience with Via Epia and Asterisk. Will this mix work in appropriate way ;) ? There is not only 1 makefile where you have to define i586. ___ --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] Via Epia
See http://www.epiawiki.org/wiki/tiki-index.php?page=EpiaInstallingGentoo regarding CFLAGS settings for different VIA CPUs. On Sat, 2005-12-10 at 11:57 +, Roger Hill wrote: Aha! I was getting the same error and could not figure out why. My CPU is a VIA Samuel. So it's a VIA thing?? Roger Andrew Nowrot wrote: Hi, Does anyone has some experience in installing * on Via Epia. I am struggling with it for about two days. And when I finally managed to install asterisk 1.0.9 after starting it I get this error or whatever: - Illegal instruction I changed the variable in makefile to i586 (I also tried i686 because that is what my uname - m says) but still I get the same problem. I use Debian with 2.4.30 kernel. Does anyone has some experience with Via Epia and Asterisk. Will this mix work in appropriate way ;) ? Cheers Andrew ___ --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] GotoIf always goes to true?
Shouldn't the _expression_ be
GotoIf($[${T}3]?3:7)
On Fri, 2005-11-18 at 15:39 -0800, Andy Kuo wrote:
Hi all,
I recently found GotoIf not working right in my extensions.conf, so I write a simple test and test it on my newly installed v1.2 box.
However, in all cases, GotoIf seems to always result in true. This happens to me in both ABE and V1.2
my extensions.conf :
[globals]
Music=123
[default]
exten = ${Music},1,Answer
exten = ${Music},2,SetVar(t=1)
exten = ${Music},3,NoOp(${TIMESTAMP} - ${T})
exten = ${Music},4,MP3Player(/var/lib/asterisk/mohmp3/deck.mp3)
exten = ${Music},5,SetVar(t=$[${T} + 1])
exten = ${Music},6,GotoIf(${T}3?3:7)
exten = ${Music},7,Hangup
CLI output :
*CLI -- Executing Answer(SIP/100-274e, ) in new stack
-- Executing Set(SIP/100-274e, t=1) in new stack
-- Executing NoOp(SIP/100-274e, 20051118-153136 - 1) in new stack
-- Executing MP3Player(SIP/100-274e, /var/lib/asterisk/mohmp3/deck.mp3)
in new stack
Nov 18 15:31:39 NOTICE[5414]: app_mp3.c:108 timed_read: Poll timed out/errored o
ut with 0
-- Executing Set(SIP/100-274e, t=2) in new stack
-- Executing GotoIf(SIP/100-274e, 23?3:7) in new stack
-- Goto (default,123,3)
-- Executing NoOp(SIP/100-274e, 20051118-153139 - 2) in new stack
-- Executing MP3Player(SIP/100-274e, /var/lib/asterisk/mohmp3/deck.mp3)
in new stack
Nov 18 15:31:42 NOTICE[5414]: app_mp3.c:108 timed_read: Poll timed out/errored o
ut with 0
-- Executing Set(SIP/100-274e, t=3) in new stack
-- Executing GotoIf(SIP/100-274e, 33?3:7) in new stack
-- Goto (default,123,3)
-- Executing NoOp(SIP/100-274e, 20051118-153142 - 3) in new stack
-- Executing MP3Player(SIP/100-274e, /var/lib/asterisk/mohmp3/deck.mp3)
in new stack
Nov 18 15:31:45 NOTICE[5414]: app_mp3.c:108 timed_read: Poll timed out/errored o
ut with 0
-- Executing Set(SIP/100-274e, t=4) in new stack
-- Executing GotoIf(SIP/100-274e, 43?3:7) in new stack
-- Goto (default,123,3)
-- Executing NoOp(SIP/100-274e, 20051118-153145 - 4) in new stack
-- Executing MP3Player(SIP/100-274e, /var/lib/asterisk/mohmp3/deck.mp3)
in new stack
Nov 18 15:31:48 NOTICE[5414]: app_mp3.c:108 timed_read: Poll timed out/errored o
ut with 0
I have been trying to figure this out for the past few days. I think it must be some stupid mistake of mine, but just can't figure out what/where.
Please help.
Thank you very much.
Andy
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Re: [Asterisk-Users] Asterisk 1.2 Released!
cd top level asterisk source directory (where UPGRADE.txt is) patch -p0 /path/to/silence-suppression-2.diff On Thu, 2005-11-17 at 07:07 -0500, Asterisk guy wrote: does the following patch work for 1.2? how to apply it to 1.2? ( I am not a programmer, don't know how to use .diff file). http://bugs.digium.com/view.php?id=5374 silence-suppression-2.diff On 11/17/05, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Asterisk guy wrote: does it include the patch for VAD? ( dropping extra frame of G.729 since we already have a VAD frame at the end ) It does not include several important things. It does not include a SIP jitter buffer. It does not include the ability to use Zaptel for timing of the RTP audio. It does not include VAD/CND support. As far as I know it also does not have the patch to make the new IAX2 jitterbuffer work correctly when connecting to a 1.0.x server. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropping extra frame of G.729 since we already have a VAD frame at the end
On Wed, 2005-11-16 at 19:31 +1300, Matt Riddell wrote: Sergey Okhapkin wrote: Already supported (simple patch exists). http://bugs.digium.com/view.php?id=5374 Which? silence-suppression-2.diff It's already supported in Asterisk or you can patch Asterisk to add it? You can patch Asterisk (CVS HEAD). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] dropping extra frame of G.729 since we already have a VAD frame at the end
Already supported (simple patch exists). http://bugs.digium.com/view.php?id=5374 On Wed, 2005-11-16 at 12:32 +1300, Matt Riddell wrote: Asterisk guy wrote: dropping extra frame of G.729 since we already have a VAD frame at the end- Turn off VAD, it is not supported by Asterisk. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy echo?
Lower speaker volume on the phone connected to IAXy. On Mon, 2005-11-14 at 07:21 -0600, Mike Hammett wrote: I've got two customers on the same broadband provider. Same Asterisk box on my end. Same CLEC. One has an IAXy and the other has an Asterisk box with an array of devices (Grandstream, Cisco, ATCOM, xten, etc.). The people behind the Asterisk box have had no audio quality issues. The person with the IAXy often encounters an echo. The echo is only heard on the remote side and it only contains the remote caller's voice. This echo has been heard with the remote side being varying LECs. The echo is not always there. I'd almost say that the echo is not there more than it is. Troubleshooting next step? I haven't changed out the IAXy because I don't have any other ATAs to put in place. Mike Hammett Intelligent Computing Solutions http://www.ics-il.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Comments in AEL files?
//comment AEL ignores any text from // till the line end. On Mon, 2005-11-14 at 07:39 -0800, Ed Greenberg wrote: Any way to comment out a line (or some text) in an AEL file? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk realtime extensions context inclusion
That's what macro is useful for. Don't include these common contexts, but convert them to macros and call these macros from user's dialplan. Macros in asterisk dialplan are close to subroutines rather than to C-style #define. On Mon, 2005-11-14 at 07:44 +, Daniel Clark wrote: Thanks for the reply, its an approach I didnt think of to simply include the information from the other contexts into where I would be including from. In most cases that would work, but not in my case. Each user of my system will be able to place outgoing calls using their own sip connection (as in one they create with sipgate or vonage etc). To ensure that each user can dial out with their own sip connection and nobody elses they are each getting their own context and that context is the only place in the dialplan to dial that particular external sip connection. For a small amount of users its possible to include all the information in each context, however Im dealing with 15,000 users and would like a database small enough to fit on the hard disk! Would it not be possible to do something with the Goto app? In each persons dialplan I can have an extension to catch internal numbers and then forward to another context using exten = 1,1,Goto(context2) or something like that? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WARNING[3035]: Invalid priority/label ' ' at line 17
Post line 17 of your extensions.conf file. On Sat, 2005-11-12 at 18:47 -0600, Greg Blakely wrote: I just recently upgraded to the latest HEAD, and am now getting the following warning: -- Including context 'fromcnet' in context 'pots' Nov 12 18:45:17 WARNING[3035]: pbx_config.c:1697 pbx_load_module: Invalid priority/label '' at line 17 -- Including context 'longdistance' in context 'international' I have added a comment line above and below every config file that I have in /etc/asterisk, and the warning never changes. What's up with this? And will it affect anything? TIA ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Result branching in AEL
n+101 feature is deprecated and is no longer supported in Asterisk.
All applications are modified to set exit status variable. Use something
like
VoiceMail(b${EXTEN});
if(${VMSTATUS} = FAILED) {
Noop(mailbox doesn't exists);
}
On Fri, 2005-11-11 at 10:11 +, Chris Bagnall wrote:
Morning all,
I'm trying to rewrite my dialplan macros into AEL. How does one handle
result-dependent branching (e.g. VoiceMail will branch to n+101 if mailbox
doesn't exist) in AEL? Or is there a better way of doing this?
Thanks in advance.
Regards,
Chris
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Re: [Asterisk-Users] IM / presence asterisk-1.2-RC1
Asterisk sends OPTIONS message if the device have qualify=NNN option set. On Fri, 2005-11-11 at 11:24 +0100, harry gaillac wrote: Here are some other files. Why asterisk send sip OPTION message to agents ? Harry 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x81cf940 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted Retransmitting #2 (NAT) to 192.168.0.20:5060: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.119.11.222:5060;branch=z9hG4bK4a119599;rport From: asterisk sip:[EMAIL PROTECTED];tag=as747a6ef0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Nov 2005 10:23:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8194ea0 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted /// --- harry gaillac [EMAIL PROTECTED] a écrit : Sorry, Here are some files Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=1000 ; Default length of incoming/outoing registration allow=all ; First disallow all codecs musicclass=default ; Sets the default music on hold class for all SIP calls language=fr ; Default language setting for all users/peers rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity tpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=Asterisk PBX ; Allows you to change the user
Re: [Asterisk-Users] Choppy Audio in Echo Test and Music On Hold (1.2.0-b2)
Do you have silence suppression enabled on your clients? Asterisk can't work with silence suppression. Take a look at http://bugs.digium.com/view.php?id=5374 , the patch works fine to me, now I'm able to set SS and save bandwidth. On Mon, 2005-11-07 at 19:41 -0800, Chris Tracy wrote: I recently resurrected an old athlon system and put CentOS 4.2 on it to play with asterisk. First I tried asterisk-1.0.9, now I'm using 1.2.0-b2. Both have the same audio issues that have me stumped. I looked through all the lists and forums and the closest I could get were some messages from 2003: http://lists.digium.com/pipermail/asterisk-users/2003-August/017171.html I've got asterisk set up with my xten-lite softphone on extension 200 over SIP. I've configured extension 611 as an echo test and 612 will play 30 seconds of MusicOnHold. I can connect to both just fine, however, they sound rather bad when they work (quite muddy) and periodically they just drop out for as much as 5 seconds before coming back. Enabling all the debugging and verbosity options, I've found a few messages that occur during each drop. During the MOH run, every time there's a drop, the console scrolls: res_musiconhold.c:535 monmp3thread: Only wrote -1 of 640 bytes to pipe over and over until the sound comes back, at which point, the console message: rtp.c:1247 ast_rtp_raw_write: Difference is 33824, ms is 4248 is displayed. (Not always the same numbers in that one, obviously) In the echo test, again, after a drop, the audio returns and a message similar to: rtp.c:1247 ast_rtp_raw_write: Difference is 12496, ms is 1582 is displayed. The asterisk server is on a single Athlon MP 1600+ (1.4GHz) with 512MB of RAM. It's got a K7D-Master mobo, and is connected to the system running the softphone through a 100Mbit LAN. I've not enabled any of the MMX optimizations as there were warnings that they didn't play nice with AMD chips. If there's any further info I can provide, I'd be happy to. Thanks, Chris ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Caller ID How does it get setup?
AFAIK, most of VOIP providers ignore callerid from ATA and substitute it with a caller id on their records. On Fri, 2005-11-04 at 01:11 -0600, Jason Brashear wrote: OK I am exhausted. I can't seem to figure out how to send a caller ID along with a Outbound call. Can you believe that I got Vonage to reset my Cisco ATA for $15.00 I then canceled my account! Well I was with them for over two years, now I am running Asterisk like the big boys! LOL... Anyway, Outbound Caller ID Hos is this done? I now use VoicePulse as my provider. -Jason ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Invalid/Timeout handlers in ael?
As far as I (don't:-) understand asterisk sources, extensions 't' and 'i' can be used in _context_ only, but not in macro. Application voicemail has a special handling for 'a' extension to allow it in macro.
On Thu, 2005-11-03 at 16:32 -0800, John Biundo wrote:
Does anybody know how to code invalid and timeout handlers in ael macros?
I tried the following, but no luck.
=
macro call-screen() {
NoOp(Macro call-screen);
Background(privacy-screening-unidentified-calls);
tryagain:
Playback(pls-rcrd-name-at-tone);
Set(SCREEN_FILE=/tmp/screen-${EPOCH});
Record(${SCREEN_FILE}.wav,6,25);
catch i {
NoOp(invalid?);
goto tryagain;
}
catch t {
NoOp(timeout?);
goto tryagain;
}
};
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RE: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana
AFAIK, the official language of this mailing list is English. On Tue, 2005-11-01 at 08:54 -0500, Carlos Alperin wrote: Walter, No se acerca de que es lo mas atractivo. El servicio puede ser en el horario que tu quieras (Nosotros trabajamos 24 hs, 7 dias a la semana, 365 dias al ao), pero si quieres restriccin de horario, se puede hacer. No dije nada acerca del espaol Carlos From:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Walter Willis Sent: Monday, October 31, 2005 7:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana 2005/10/31, Walter Willis [EMAIL PROTECTED]: Tengo un asterisk en al oficina de un cliente , que quiere hacer llamadas ilimitadas a estados unidos; las llamadas tienen que ser al mismo tiempo. alguien ofrecio una conexion iax2 para los 4 usuarios. no tienes algo mas atractivo se supone que funcionaria eso desde las 3:00pm hasta las 8:00pm diariamente ecepto los domingos. cuanto costaria eso. y gracias por als respuestas. noes que el espaol sea malo sino que el teclado era malo y aparte el tiempo era corto. 2005/10/31, Manny A. Wise [EMAIL PROTECTED]: NO estoy para juegos, te conteste y te ofreci lo que pediste, cual es tu problema? -Original Message- From: Walter Willis [mailto:[EMAIL PROTECTED]] Sent: Monday, October 31, 2005 7:05 PM To: Manny A. Wise Subject: Re: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana pense que ibas a decir algo mas interesante, parece que estas probando tu correo en la lista. 2005/10/31, Manny A. Wise [EMAIL PROTECTED]: Y si estas en el Peru y hablas espanol, por que no lo escribes correctamente? -Original Message- From: Walter Willis [mailto:[EMAIL PROTECTED]] Sent: Monday, October 31, 2005 4:50 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana you provider tha service unlimited call usa i want the trunk with 4 users to unlimited USA iax or iax2 2005/10/31, Manny A. Wise [EMAIL PROTECTED]: Do you speak English? Your Spanish is bad too.. Puedo ayudarte en espanol Yo puedo link tu asterisk a my asterisk Manny -Original Message- From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Walter Willis Sent: Monday, October 31, 2005 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] llamdas por 4 lineas a eeuu tarifa plana tengo un asterisk, alguien conoce algun proveedor que brinde el sistema de linkar mi asterisk a su servicio para tener tarifa plana a eeuu. para llamar por 4 conexiones al miamo tiempo desde mi asterisk? me parece haber visto que se configuraba con una troncal iax2 2005/10/31, [EMAIL PROTECTED] [EMAIL PROTECTED]: Does anyone know where the official release of Asterisk 1.2 is? Do we have a time-frame of when this version will be released and how much longer it will be in BETA. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk+Nat+sipura (Help)
I don't think the problem is NAT-related. Looks like To header in SIP INVITE message do not match to User ID in sipura settings. On Thu, 2005-10-27 at 01:57 +0330, mohammad mirzaee wrote: Hi ALL; I have users with Sipura/Linksysphones regsitered behind Nat( useing STUNat phonenot portforwarding) in my Asterisk box, when I try to call them with another phone i got: Got SIP response 404 Not Found back from 217.6.190.4 SIP/217.6.190.4:5060-666d is circuit-busy Isabove mentioned problem relates to Nat, Is there anybody who use sipura with STUN method and can recive calls? My asterisk Sip.conf for Nat has the following: [sipura] .. nat=yes canreinvite=no qualify=1000 Appreciate any help Mohammad ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Format of extensions.conf
On Tue, 2005-10-25 at 00:52 -0400, Leif Madsen wrote: Now, as someone has also pointed out, using quotes around the string is probably better form as it should handle spaces and such. In expressions only. Set() command is broken in this area (1.2beta and CVS HEAD). To clear, for example, calleridname one must write Set(CALLERID(name)=) The command Set(CALLERID(name)=) will set the name part of callerid to guess what?-) Yes, to a string containing 2 double quote characters! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Format of extensions.conf
Taking in the account poor Asterisk documentation, it's a bug. The bug can be called as a feature, only when it is documented:-) On Tue, 2005-10-25 at 10:47 -0400, Leif Madsen wrote: Set(CALLERID(name)=) will set the name part of callerid to guess what?-) Yes, to a string containing 2 double quote characters! Yes, I was speaking of expressions specifically, but thanks for clearing this up. Honestly, I don't think I've ever tried to set a NULL string to a variable with a function... that could probably be filed as a bug, but someone might call it a feature :) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with asterisk 1.0.9 and sip and dtmf
Check if you have Silence Suppression disabled on PSTN line of spa-3000 (admin/advansed/PSTN line). On Sat, 2005-10-22 at 18:05 -0400, Mike Bernson wrote: I have asterisk running with sipura 3000 connect to PSTN and sipura 2000 connected to phones. On inbound calls I am getting what sounds like DTMF tone when someone is talking on the PSTN side of the phone. It sound like someone is hitting key on the phone while talking. Is there any way to stop this from happing. Here is the PSTN and one ext from the sip.conf PSTN line [199] username= type=friend secret= record_out=Adhoc record_in=Adhoc qualify=yes port=5061 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=incomming canreinvite=no ext [206] username= type=friend secret= record_out=Adhoc record_in=Adhoc qualify=yes port=5061 nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 context=from-sip canreinvite=no ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does fwdout even work anymore?
I just tried to place a call thru fwdout, works fine. On Sat, 2005-10-22 at 06:40 -0700, trixter aka Bret McDanel wrote: Initially I thought this may have been the fiasco last night or the night before (I forget now) where level3 did a software upgrade and it went awry. With the pings responding I now wonder. It still could be this (all symptoms from the same problem). I am thinking about signing up for FWD-out anyway, I might do that tonight and see if it works for me. I dont know the exact routes that everyone is using to get there, which would play a role in this. Just flinging wild guesses based on current events. On Sat, 2005-10-22 at 08:21 -0600, Rich Adamson wrote: Mine stopped working sometime back in Feb. I just made the changes so everything points to fwdOUT.net now, but it still seems to fail. Using a sniffer, I see packets going out, but none coming back. I have a firewall, but 4569 has been opened, and I'm not seeing denys on the firewall anyway. I'm just not getting a response. Any ideas? ~jay FWD used work not to long ago, but is not working today. IAX registration to FWD is not going through. Is anybody lucky? As of 8:20 am CDT, both FWD and Iaxtel.com are unresponsive. It appears the FWD iax server can be reached via a ping, but there is no response from it for a iax register. That would imply their asterisk crashed but the server is up. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why Asterisk documentation is so poor...
http://bugs.digium.com/view.php?id=5472 The users will not learn about undocumented AEL features. Sure I'm not going to reopen the problem. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why Asterisk documentation is so poor...
I made a lot of contributions to many open source projects already, I never saw such pressure from the code maintainers to code contributors, usually it's up to maintainers how to apply the changes proposed by the contributor. I put a note that you can rephrase as you wish to follow asterisk's maintainers roadmap and guidelines. I'm not fluent in english also, to express your wishes in the way you want. On Thu, 2005-10-20 at 15:50 +0200, Olle E. Johansson wrote: Sergey Okhapkin wrote: http://bugs.digium.com/view.php?id=5472 The users will not learn about undocumented AEL features. Sure I'm not going to reopen the problem. Sergey, I am sorry if you took our comments that badly. I proposed a worthing and you did not accept that and refused to update according to our suggestions. Tilghman therefor decided to close the bug. I suggest you try again, re-open the bug, fix the problem and continue to add more documentation. We do need more documentation! It has to be correct though, and that's why we are giving feedback. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_odbc with tds
Looks like a bug to me, I just took a look at cdr_odbc.c. I'm not a specialist in ODBC, but what seems to me wrong is the module does INSERT into the database, but does not make COMMIT. On Thu, 2005-10-20 at 16:31 +0100, Ben merrills wrote: Does anyone know why, using latest cvs head, freetds 0.62.1-0 and unixODBC, when running cdr_odbc, it says it's logged the call successfully, however, when checking the table, nothing is there! I checked through the bug tracker; and a problem very much like mine was in there, with status resolved as of last year (1339). Can anyone shed some light on this please? Cheers, Ben ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] cdr_odbc with tds
...Or fix the problem yourself:-) On Thu, 2005-10-20 at 16:58 +0100, Ben merrills wrote: What should I do? :) Add it to the bug tracker? Ben -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergey Okhapkin Sent: 20 October 2005 16:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] cdr_odbc with tds Looks like a bug to me, I just took a look at cdr_odbc.c. I'm not a specialist in ODBC, but what seems to me wrong is the module does INSERT into the database, but does not make COMMIT. On Thu, 2005-10-20 at 16:31 +0100, Ben merrills wrote: Does anyone know why, using latest cvs head, freetds 0.62.1-0 and unixODBC, when running cdr_odbc, it says it's logged the call successfully, however, when checking the table, nothing is there! I checked through the bug tracker; and a problem very much like mine was in there, with status resolved as of last year (1339). Can anyone shed some light on this please? Cheers, Ben ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Priority jump in AEL
There is no way in AEL to specify the priority explicitly. To solve the problem use DB_EXISTS function. Here is an example from my dialplan:
if(${DB_EXISTS(Provider/${prov}/used)})
Set(MINUTES_USED=${DB_RESULT});
On Tue, 2005-10-18 at 21:16 +, Kris Edwards wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I use this macro for call screening:
[macro-screen]
exten = s,1,Wait(1)
exten = s,2,DBget(SCREENFILE=callerid/${CALLERIDNUM})
exten =
s,3,ParkAndAnnounce(beep:beep:callfrom:${SCREENFILE}:holdingonexten:PARKED:beep:beep:${SCREENFILE}:isholdingonext:PARKED|180|${ARG1}|${ARG2})
exten = s,103,Set(SCREENFILE=/var/lib/asterisk/sounds/names/${UNIQUEID})
exten = s,104,Playback(unknownid)
exten = s,105,Record(${SCREENFILE}:gsm|3)
exten = s,106,System(/usr/bin/normalize -g 6db ${SCREENFILE})
exten = s,107,DBput(callerid/${CALLERIDNUM}=${SCREENFILE})
exten =
s,108,ParkAndAnnounce(beep:beep:callfrom:${SCREENFILE}:holdingonexten:PARKED:beep:beep:${SCREENFILE}:isholdingonext:PARKED|40|${ARG1}|${ARG2})
I'm trying to convert my dialplan to ael, but I don't get how to handle
the jump if there is no entry in the database for the caller. I'm
guessing it's an if statement, but what does the db return if there is
no entry? 0, null?? If somebody could get me started with what that
staement should be (at 103) then I should be good to go.
(If this is a stupid question or explained elsewhere, feel free to let
me know)
Thanks,
Kris
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jVxDtsvzMnjdjtj0EwMqevk=
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Re: [Asterisk-Users] DID setup from goiax.com
Replace
[goiax]
with
[PHONENUMBER]
username= don't work for users in IAX channel.
On Wed, 2005-10-19 at 10:27 -0700, [EMAIL PROTECTED] wrote:
That is What I stated in the email.. my GOIAX #. not the DID #.
That is not the issue.
for the incoming context put your goiax.com http://goiax.com phone
number
not the free DID number but the other one.
On 10/19/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Trixter:
Thanks for the guide to setting this up:... I have tried the below
configuration with my settings, and when I place /goiax-in after my
register command, my register statement fails.
If i remove it. I get a Rejected connect attempt from goiax's server IP,
trying to reach 's@'
I have put my GoIAX # in default, local, as the extension, and nothing.
I dont know where to look next on why i'm getting the rejected connect
attempt.
Thanks..
./Ben
On Wed, 2005-10-19 at 14:24 +0800, Ronald Wiplinger wrote:
Can anybody post a step by step setup guide, please?
Its like anything else once you have signed up ...
in iax.conf
register =
PHONENUMBER:[EMAIL PROTECTED]/goiax-inhttp://PHONENUMBER:[EMAIL
PROTECTED]/goiax-in
[goiax]
type = peer
host = server1.goiax.com http://server1.goiax.com
context = default
secret = PASSWORD
allow = gsm
;allow = ulaw
;disallow = all
notransfer = yes
qualify = yes
auth = md5
username = PHONENUMBER
replace PHONENUMBER with the 8782 number you were issued. Replace
PASSWORD with your password from you account signup.
Then in extensions.conf
; for outbound
exten = _1NX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,R)
exten = _1NX,2,Busy
exten = _1NX,102,Congestion
exten = _1NX,202,playback(tt-weasels)
; for inbound
exten = goiax-in,1,DO WHATEVER HERE
asterisk -rx reload
you should be set.
--
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605 Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
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Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com http://www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Phone: 845-652-4578 x205
Phone: 978-203-3848 x205
Fax: 518-631-2856
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Re: [Asterisk-Users] Free DID's
Callpacket.com has a free plan (up to 100 mins/month outbound, unlimited inbound, free DID). On Wed, 2005-10-19 at 13:31 -0700, trixter aka Bret McDanel wrote: On Wed, 2005-10-19 at 16:19 -0400, Yu Safin wrote: am I correct in believing that only goiax.com offers free DID's? nope you are not correct. ipkal.com offers (to any sip proxy now not just to FWD) washington state DIDs free stanaphone.com offers new york state DIDs free calluk.com offers UK numbers free sipgate.co.uk offers german and UK numbers free voipbuster.com offers free outbound to 14 european countries (landline only) and free to US mobiles and landlines. Just gotta put $5 on your account via paypal to get past the 1 minute call limit and get to 1 hour call limits. IF I MISSED ANY FREE PROVIDERS THAT SOMEONE ELSE KNOWS OF PLEASE REPLY TO THE LIST SO THAT A MORECOMPLETE LIST CAN BE CREATED. goiax.com offers free inbound with at least Mass DIDs and free US outbound (although that is currently suspended Matthew indicated he liked one method and that should be back up soon). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free DID's
On Wed, 2005-10-19 at 14:15 -0700, Jesse Keating wrote: On Wed, 2005-10-19 at 16:41 -0400, Sergey Okhapkin wrote: Callpacket.com has a free plan (up to 100 mins/month outbound, unlimited inbound, free DID). Do you have hints on using callpacket w/ Asterisk? register = sipusername:[EMAIL PROTECTED] [callpacket-out] type=peer username=sipusername secret=sipsecret fromuser=sipusername host=ser.callpacket.com dtmfmode=rfc2833 [callpacket-in] type=user host=ser.callpacket.com dtmfmode=rfc2833 context=from-callpacket ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Free DID's
Thanx! voip.callpacket.com didn't work before - 487 Too many Hoops (double o:-) in response to REGISTER. That's why I used IP address. Looks like they fixed something:-) On Wed, 2005-10-19 at 16:26 -0700, Thameem Ansari wrote: For callpacket host try using voip.callpacket.com. This is a recent change they made and ser.callpacket.com will not work. But if you nslookup both the names pointing to same ip. They might be using some kind of virtual hosting on that name I think. -Thameem On 10/19/05, Sergey Okhapkin [EMAIL PROTECTED] wrote: On Wed, 2005-10-19 at 14:15 -0700, Jesse Keating wrote: On Wed, 2005-10-19 at 16:41 -0400, Sergey Okhapkin wrote: Callpacket.com has a free plan (up to 100 mins/month outbound, unlimited inbound, free DID). Do you have hints on using callpacket w/ Asterisk? register = sipusername:[EMAIL PROTECTED] [callpacket-out] type=peer username=sipusername secret=sipsecret fromuser=sipusername host=ser.callpacket.com dtmfmode=rfc2833 [callpacket-in] type=user host=ser.callpacket.com dtmfmode=rfc2833 context=from-callpacket ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] goiax configuration help please
Replace [goiax] with [87820myid]. Just replace the section name.
On Wed, 2005-10-19 at 20:21 -0400, Jim Duda wrote:
I saw the posting concerning goiax offering free DIDs. I went ahead,
created an account, and got myself a DID.
Who is goiax, and how can they be doing this for free? It's nice, but
how can they offer that?
I have outbound calling working from asterisk, to 800 numbers.
I cannot seem to get inbound calls working though. I cannot figure out
why. I get a message from asterisk saying that asterisk rejected the
call due to an authorization failure. Asterisk reported a failure at a
specific line of code in chan_iax2.c and it has to do with
authentication. I'm registered with goiax as I see the proper result in
iax2 show registry. So, I at least have the correct ID and SECRET.
Do these DIDs really work?
In my iax.conf, I have:
;
; GOIAX
;
register = 87820myid:mysecret@server1.goiax.com
[goiax]
context=home
type=friend
host=server1.goiax.com
auth=md5
username=87820myid
secret=mysecret
disallow=all
allow=ulaw
allow=gsm
In my extensions.conf, I have (2 entries since I wasn't sure which
number would be used)
;
; Goiax
;
exten = mydid,1,AGI(MisterHouse.agi,CallerID)
exten = mydid,2,Dial(${PHONES0}${PHONES1}${PHONES2},20,tr)
exten = mydid,3,Macro(voicemail,${PHONES0VM})
exten = mydid,4,Hangup
exten = mydid,103,Macro(voicemail,${PHONES0VM})
exten = mydid,104,Hangup
exten = 87820myid,1,AGI(MisterHouse.agi,CallerID)
exten = 87820myid,2,Dial(${PHONES0}${PHONES1}${PHONES2},20,tr)
exten = 87820myid,3,Macro(voicemail,${PHONES0VM})
exten = 87820myid,4,Hangup
exten = 87820myid,103,Macro(voicemail,${PHONES0VM})
exten = 87820myid,104,Hangup
Can anyone see what I might be doing wrong?
Jim
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Re: RE:[Asterisk-Users] free dids on goiax.com
I completely agree. No reason to provide unlimited free service, put some reasonable restrictions like no more 10 different numbers could be called a day or no more than 20 calls a day. On Tue, 2005-10-18 at 17:39 -0500, Rajesh kumar wrote: For a free service, its quite acceptable to demand certain conditions and regulations just to keep the abusers away. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] free dids on goiax.com
I expect all 175 DIDs has gone already... On Tue, 2005-10-18 at 17:58 -0700, John Wenger wrote: On 10/18/05, Matthew Simpson [EMAIL PROTECTED] wrote: GoIAX, the Asterisk community's free IAX provider, is offering free US dids now.I loaded about 175 dids in and put up a very beta sign in page. snip. yours, Matthew Simpson GoIAX -- www.goiax.com TxLink -- www.txlink.net _ Perhaps there is a problem with my Firefox browser, but I can not see my DID when I log in, and I can not register for a DID. I see: -- You currently have DID assigned. (add remove option here) Available DIDs: - but there is nothing to be seen when I click on the little arrow to see what is available. Clicking on the Assign DID button causes a reminder to choose a DID first, e.g., Form error. How did you get here without choosing a DID? Has anyone got this signup to work with a linux-based browswer? John ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
Are the devices at 200 and 310 set up to register with your asterisk? On Sat, 2005-10-15 at 11:42 -0700, Samy Antoun wrote: Hi, I have 3 SIP extensions, setup as follows: # Device Location options 200 Sipura local 210 Sipura remote nat=yes qualify=yes 310 eyebeam remote nat=yes qualify=yes This is the result of sip show peers: Name/user Host Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored Does anyone know why ext 210 the only one has a ping status OK (305 ms) and the others are Unmonitored Regards __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
Hmm.. What is the output of sip show users and sip show peers? On Sat, 2005-10-15 at 12:30 -0700, Samy Antoun wrote: --- Sergey Okhapkin [EMAIL PROTECTED] wrote: Are the devices at 200 and 310 set up to register with your asterisk? Yes, they are registered and I can call them __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Outbound registration expirey
The host you're registering with sets the expiry parameter to 60 seconds in the reply message. Use sip debug to see SIP messages running. On Fri, 2005-10-14 at 17:00 -0300, Ricardo Poppi wrote: Hi list! Im connecting a Brasilian voip (- gvt.com.br -) provider through my asterisk box and using the register command from sip.conf. What I cant understand is why my unit sends a new registration message every minute! And every time my asterisk box sends a registration, it gots a sucessful response, and shows de message: Oct 14 16:48:22 NOTICE[4090]: chan_sip.c:8742 handle_response_register: Outbound Registration: Expiry for gvt.com.br is 60 sec (Scheduling reregistration in 45 s) Im using asterisk-1.2.0-beta and the sip.conf parameters about registration: defaultexpirey=1200 registertimeout=1200 There is any way to make asterisk follow the 1200 seconds Im trying to tell? Could be something happening out of my unit but at the provider network? Thanks in advance, Ricardo Poppi. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Starting simple switch from an extension?
DISA(password|context) On Thu, 2005-10-13 at 12:58 +0100, Derek Conniffe wrote: Hi, Is there a command to start simpleswitch from an extension? For example it would allow me to dial in to my * box and get a dial tone to make an outgoing call. Thanks, Derek ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
