Re: [asterisk-users] AGI: Not getting answers from get_data in a call-file call
> Have you tried with AGI Debug on?
Yes! Even before you asked :-)
This is when I use DeadAgi (for some reason):
-- Executing [EMAIL PROTECTED]:3] DeadAGI("Zap/4:103-1", "callback2.agi")
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/callback2.agi
AGI Tx >> agi_request: callback2.agi
AGI Tx >> agi_channel: Zap/4:103-1
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Zap
AGI Tx >> agi_uniqueid: 1202098745.49994
AGI Tx >> agi_callerid: 0288159096
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 3
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 33
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 88159305
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: barnet-callback
AGI Tx >> agi_extension: h
AGI Tx >> agi_priority: 3
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << CHANNEL STATUS
AGI Tx >> 200 result=6
AGI Rx << SAY DIGITS 6 ""
[Feb 4 15:19:19] WARNING[19954]: file.c:643 ast_readaudio_callback: Failed to
write frame
-- Playing 'digits/6' (language 'en')
AGI Tx >> 200 result=-1
AGI Rx << GET DATA one-moment-please 1 3
[Feb 4 15:19:19] WARNING[19954]: file.c:643 ast_readaudio_callback: Failed to
write frame
-- Playing 'one-moment-please' (language 'en')
AGI Tx >> 200 result=-1
AGI Rx << SAY DIGITS 1 ""
[Feb 4 15:19:19] WARNING[19954]: file.c:643 ast_readaudio_callback: Failed to
write frame
-- Playing 'digits/1' (language 'en')
AGI Tx >> 200 result=-1
-- AGI Script callback2.agi completed, returning -1
And this is with normal AGI:
-- Executing [EMAIL PROTECTED]:3] AGI("Zap/4:100-1", "callback2.agi") in
new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/callback2.agi
AGI Tx >> agi_request: callback2.agi
AGI Tx >> agi_channel: Zap/4:100-1
AGI Tx >> agi_language: en
AGI Tx >> agi_type: Zap
AGI Tx >> agi_uniqueid: 1202099898.50183
AGI Tx >> agi_callerid: 0288159096
AGI Tx >> agi_calleridname: unknown
AGI Tx >> agi_callingpres: 3
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 33
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 82572599
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: barnet-callback
AGI Tx >> agi_extension: 0288159096
AGI Tx >> agi_priority: 3
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
AGI Rx << ANSWER
AGI Tx >> 200 result=0
AGI Rx << CHANNEL STATUS
AGI Tx >> 200 result=6
AGI Rx << SAY DIGITS 6 ""
-- Playing 'digits/6' (language 'en')
AGI Tx >> 200 result=0
AGI Rx << GET DATA one-moment-please 1 3
-- Playing 'one-moment-please' (language 'en')
AGI Tx >> 200 result= (timeout)
-- AGI Script callback2.agi completed, returning 0
When running it as a normal call (i.e. not initated by a call-file),
it shows up with:
AGI Rx << GET DATA one-moment-please 1 3
-- Playing 'one-moment-please' (language 'en')
AGI Tx >> 200 result=354
AGI Rx << SAY DIGITS 354 ""
-- Playing 'digits/3' (language 'en')
-- Playing 'digits/5' (language 'en')
-- Playing 'digits/4' (language 'en')
AGI Tx >> 200 result=0
--
Edwin Groothuis |Personal website: http://www.mavetju.org
[EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/
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Re: [asterisk-users] AGI: Not getting answers from get_data in a call-file call
> a) The call file Channel: Zap/g4/0409227633 MaxRetries: 0 RetryTime: 60 WaitTime: 30 Extension: 0409227633 Callerid: 0409227633 Context: barnet-callback Priority: 1 > b) the snippet of extensions.conf that this is called in ; ; dial back ; exten => 0293353699,1,AGI(callback1.agi) [barnet-callback] exten => _.X,1,NoOp(callback time) exten => _.X,n,Answer() exten => _.X,n,AGI(callback2.agi) exten => _.X,n,Hangup exten => OutgoingSpoolFailed,1,NoOp(Failed) You dial in on 02 9335 3699, come into the AGI script which creates the above call-file (that works), it calls you back on via the call-file and then drops into the callback2.agi script which is the one I had in the earlier email. As I said earlier: nothing strange, nothing spectacular. Just strange that it doesn't work for calls initiated by Asterisk. -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI: Not getting answers from get_data in a call-file call
I have the following situation: I drop a call-file into the Asterisk
spool directory and I get called back. That all works.
And I have this script:
#!/usr/bin/perl -w
use Asterisk::AGI;
my $AGI = new Asterisk::AGI;
my %input = $AGI->ReadParse();
$AGI->answer();
my $i;
$i = $AGI->channel_status();
$AGI->say_digits($i);
$i = $AGI->get_data("one-moment-please", 1, 3);
$AGI->say_digits($i);
As you can see, nothing serious. When running this script in a
normal telephone call, it works. When running this script in the
call created with the call-file, I do hear the output of the first
say_digits and the one-moment-please, but the pressing of the DTMF
keys is not recognized by the system, the get_data() times out and
the function returns nothing.
This script once worked in 1.2.x (It was part of a bigger project)
but now that I want to move it to 1.4 it gives this strange behaviour.
Is there anybody with a hint on how to resolve this?
Edwin
--
Edwin Groothuis |Personal website: http://www.mavetju.org
[EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/
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[asterisk-users] Maximum retries for seqno 102 when re-inviting.
We have an interesting issue: One of our providers has two softswitches. Calls coming from the first one are handled fine by asterisk, calls coming from the second one and going through the first one are euhm... dropped half a second into the RTP stream. I have opened a ticket at Digium for it: http://bugs.digium.com/view.php?id=10449 The output of "sip debug" is funny from line 366, where it is transmitting and re-transmitting a lot of re-invites back to the softswitch with CSeq 102. Has somebody else seen this behaviour, and know how to resolve it? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RAD IPmux8
Hi, I'm looking for somebody who has managed to get their IPmux8 or IPmux11 talking to an Asterisk machine. I have it setup properly I think (the two IPmux's are talking to each other, and the zttool says that the PRI is acting okay, but I'm flooded with HDLC aborts and FCS problems. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://www.mavetju.org/weblog/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Tipping Point IPS blocking Asterisk SIP quaility messages
Hi guys, Just wanted to give you a heads up, so you don't end up chasing strange issues... Since early this morning, our Tipping Point IPS is blocking the Asterisk generated SIP Quality messages (the ones which tell you how good or badly reachably a remote SIP server is) Rule 5051: SIP: PROTOS Test Suite INVITE Test Case This filter detects a test case from the PROTOS SIP testing suite. PROTOS test suites are designed to "fizz" popular protocols to discover weaknesses in particular implementation. The PROTOS SIP test suite fuzzes SIP INVITE messages by sending several thousand combinations of illegal, abnormal, and overlong values for a variety of SIP INVITE message parameters. The results of these results range from unexpected responses to denial of service conditions to classic buffer overvlow error conditions. Vendor Site: http://.eee.oulu.fi/research/ouspq/protos/ It seems to be "default to block", which will cause a couple of issues for people today :-) Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 4
On Tue, Jan 02, 2007 at 03:17:35PM -0700, [EMAIL PROTECTED] wrote: > Has anyone made this combination work together? I've tried everything > and can't seem to get it work right. It all compiles fine, but when > rxfax is called, I get an unknown symbol error. From my reading, > everything points to me having multiple copies of spandsp and it's maybe > calling the wrong one. After the complete compile of asterisk, I jump into the apps/ directory and do this: [~/asterisk/1.4/apps] [EMAIL PROTECTED]>gcc -o app_rxfax.so -shared -Xlinker -x app_rxfax.o -lspandsp After that, with ldd on app_rxfax.so you can confirm that is is being linked. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HP DL380 and the TE4xxP cards
Hello, Does anybody have experience with the Quad T1/E1 PRI cards in an HP DL380? Just a "yes it works fine" or a "never again" is enough :-) Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --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] Remember the incoming context?
On Tue, Apr 18, 2006 at 06:57:32PM -0700, [EMAIL PROTECTED] wrote: > If the number is unique in your configuration, and the context is > already included somehow in your default-context, you can just do a > Local/number, give it a try The issue is, I don't want it in the default context (which is the standard one if you omit the @), I want it in the current context. That way I can be sure that if somebody calls in the from the PSTN or the internet context, they won't end up calling out again, while if I'm calling in from one of my phone contextes, I can do the magic dial(Local/) in my macro, but then require the right context. Looks like I have to submit a patch! Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Remember the incoming context?
Greetings, Somewhere on my asterisk system, a calls come in in a certain context, for example, from-sip or from-pstn. Then the calls gets routed through the dialplan, and a macro gets called, and another one and then the call needs to be redirected to another number in the same initial context. And you can use Dial(Local/number/initialcontext) for that. Oops, this initial context is lost somewhere on the line. Unless I'm very mistaken, there is no way to find out what the original context was, is there? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Blocked channels, according to our telco... leading to CONGESTION status
Greetings, Our telco called last week, saying that a lot of channels on our PRIs are blocked. And with blocked they have the following description in the Siemens exchanges: BBAC BLOCKED BACKWARD This status is set when the partner exchange has a blocking set and the signaling of the trunk (non-CCS7) is able to report this blocking in the backward direction. This status can remain a long time. Other usages of this status are the DCMS specific blockings: "DCMS Out of Service", "Pilot Alarm" or "Analog Carrier Failure". This status may also be set for a short time when performing DIU tests. In that case it prevents outgoing seizures. BBAC BLOCKED BACKWARD The GP reports the subscriber port with BBAC if a subscriber goes off-hook and does not dial or if a subscriber does not go off-hook during a set-up connenction. After further three minutes the status of this port is changed to BPRM. BPRM BLOCKED PERMANENT When a blocking is set in the partner exchange, but the signaling of the trunk (non-CCS7) is not able to report this blocking, the trunk will be idle in the own exchange. When an outgoing seizure is generated on such a trunk, this seizure will remain without answer and the release will be unsuccessful. In that case, this status will be set. He gave me a list of all blocked channels (1 to 5 was normal busy, 6 to 20 was idle and 21 to 31 were blocked). I ran a "zap show channel zap/" on them, but couldn't find anything in the output which made me go "All idle/busy channels have this, and all blocked ones have that". When asterisk is trying to call out via one of these blocked channels, it returns the dreaded CONGESTION status. Of course, reloading the (TE405/TE410) card drivers and zaptel drivers unlocked everything, but nothing in the asterisk CLI did the same. I'm very confused about this one, and am stuck with it for some time now. If anybody has an idea on what to do (except filing a bug with Digium) or a hint what I could/should try, let me know! Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Incoming Calls Getting Crossed - Weird
> Hey, I got a weird one for you guys, I am running vanilla 1.2.4 and > have all incoming calls come in as SIP from teliax. Twice over the past > week 2 callers who have called in around the same time end up talking to > each other instead of going through the ivr or at some point during the > IVR. One said, yeah i was talking to another patient and we had a > convo. I have double checked the dialplan and the logs and everything > looks ok. Is this a possible bug or can someone tell me what i might be > missing? Its very odd but luckily fairly rare so far, i am worried it > could get worse though. I have something similar with PRI -> PRI and PRI -> SIP calls http://bugs.digium.com/view.php?id=6502 Nothing heard from Digium Support yet. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Update to the latest zaptel driver - Congestion gone, but scary write errors replaced it
Hi, Yesterday I updated asterisk to the latest zaptel driver and today my congestion problems are gone... (see http://bugs.digium.com/view.php?id=6509), only to be replaced by: Feb 17 10:02:37 DEBUG[19225] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 26 Feb 17 10:03:08 DEBUG[19274] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 216 Feb 17 10:03:34 DEBUG[19334] chan_zap.c: Write returned -1 (Resource temporarily unavailable) on channel 2 That is on zap-channels. What is going on? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: dual TE410, both span 3 is broken (Josh Krueger)
On Sun, Feb 12, 2006 at 04:30:50PM -0700, [EMAIL PROTECTED] wrote: > I've seen a similar problem before. Span 3 was throwing errors for > (what seemed to be) no reason at all. After some testing it seemed > that the number of errors thrown on Span 3 had a relationship to the > temperature inside the servers. After installing additional cooling > the errors had reduced significantly IIRC. This was across numerous > servers each having 2 Digium quad T1/E1 cards in them. It got even funnier... After some careful log-file-reading, it seemed this started to happen after the upgrade to 1.2.x. Before it we couldn't use the the 1.0.8 and 1.0.9 version of the zaptel drivers because it couldn't get an E1 link towards a C2821 and a Vegastream 400. With the 1.2.x version it at least had a link, but all these errors on the span 3... That were "old" quad PRI cards (first generation). We ordered some "new' quad PRI cards (2nd generation) and there it all works fine with... Really really really makes me wonder about what went wrong with our setup. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dual TE410, both span 3 is broken
This afternoon I finally figured out more with regarding to a strange clock-slip problem we have on our asterisk box. We have two TE410s, in E1 mode: TE410P version c01a009b They have their own interrupts: 66: 781648298 783747388 IO-APIC-level t4xxp 233: 253890977 1311504670 IO-APIC-level t4xxp They have their full 31 channels: span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 span=7,0,0,ccs,hdb3,crc4 bchan=187-201,203-217 dchan=202 And properly configured in zapata: group = 3 context=unused switchtype=national pridialplan=unknown prilocaldialplan=unknown signalling = pri_net channel => 63-77,79-93 group = 7 context=unused switchtype=national pridialplan=unknown prilocaldialplan=unknown signalling = pri_cpe overlapdial=no channel => 187-201,203-217 And on them, both the span 3 interfaces gives huge amount of HDLC errors in the logging and clockslip-ticks on the line. Juggling the cables between other interfaces, and the then-non-span-3 is clear and the new-span-3 is full of HDLC errors and clockslips. Myself I'm going with a "this must be hardware problem", unless somebody else has an idea. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Vegastream clockslip problems
We have a Vegastream 400 connected to a digium Quad PRI card in an asterisk server, for the T.38 faxing here. Problem is that there are too many clockslips on it (and they get logged by asterisk as HDLC aborts). I've double checked the configuration on both sides, replaced the cable, tried different ports etc. It all lead to no resolution for it. Is there somebody on the list who has a Vegastream 400 working fine? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel juggling, what is it good for?
I often see this happening: - ChanIsAvail returns Zap/94 - I dial out via it. - And then "Moving call from channel 94 to channel 101" Why is it moving to channel 101? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 1.0.9 - can't get link up, 1.0.7 works fine.
Last tuesday I moved the asterisk server from 1.0.7 to 1.0.9, while leaving the zaptel drivers at 1.0.7 because it was a "lunchtime" update. This is a box with two TE405Ps in it, and all eight ports in use. Today I unloaded the 1.0.7 drivers and replaced them with 1.0.9 and oh boy... two of the 8 PRIs didn't want to come back, I got a million of FCS errors over the console and I got three new messages in /var/log/messages: Aug 21 11:32:16 Found a Wildcard: Wildcard TE410P/TE405P (1st Gen) [...] Aug 21 11:32:16 VPM: Not Present [...] Aug 21 11:35:11 HDLC Receiver overrun on channel TE4/1/3/16 (master=TE4/1/3/16) [repeated several million times] The PRIs didn't come back, pri show span showed them as Provisioned, Down, Active: PRI[3]: expected Status: Provisioned, Up, Active, got Status: Provisioned, Down, Active. PRI[7]: expected Status: Provisioned, Up, Active, got Status: Provisioned, Down, Active. I rebooted the box afterwards (just to get rid of gremlins), I rebooted the box on PRI 3 (PRI 7 is connected to the telco so no rebooting there) and it stayed the same. Unloading the drivers and going back to the 1.0.7 drivers with the 1.0.9 asterisk resolved everything and now it's working again. Who has suggestions for me on what to do? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary D-channel of span 1
On Wed, Aug 03, 2005 at 11:28:19AM -0500, [EMAIL PROTECTED] wrote: > 10. Inter-Tel AXXESS failure: HDLC Bad FCS (8) on Primary > D-channel of span 1 (Gavin Hamill) > Date: Wed, 3 Aug 2005 15:32:48 +0100 > From: Gavin Hamill <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Inter-Tel AXXESS failure: HDLC Bad FCS (8) > on Primary D-channel of span 1 > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > All the messages I've read on this are from people experiencing these errors > in quiet times - I get them as soon as I plug a port on our TE410P to an > Inter-Tel AXXESS PBX.. and I get them continuously... > > I'm just sticking an * box in between ISDN30e (we're in the UK so euroisdn) > and the PBX.. and whilst the telco ISDN30e side works like a charm [1] I > simply can't get a reliable link to the PBX.. > > I've tried two different T1 crossovers (1-4, 2-5) with identical results and > zapata.conf is indeed using signalling=pri_cpe for the telco ISDN30e and > pri_net for the PBX > > Digium support have taken me through loopback testing which came out perfect, > and the card is not sharing any IRQ, yet this error renders the card > useless :( Digium are reluctant to accept a return and replace the card since > they don't believe it to be at fault - and neither do I. > > I see the same behaviour with 1.0.9 asterisk / libpri and 1.0.9.1 zaptel... > and CVS-HEAD versions of everything. Try the 1.0.7 zaptel drivers with the 1.0.9 asterisk. Works fine here. Scary but true. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: TDM400P REV I issues - ProSLIC vs TDM400P
On Mon, Aug 01, 2005 at 11:56:26PM -0500, [EMAIL PROTECTED] wrote: > On Tue, Aug 02, 2005 at 02:48:00PM +1000, Edwin Groothuis wrote: > > The REV I card shows up in the PCI table as: > > > > 02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or > > 02:05.0 Class 0280: e159:0001) > > Subsystem: Unknown device b119:0001 > > > > But the REV E/F shows up as: > > > > 02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX > > Modem/ISDN interface (or > > 02:0d.0 Class 0780: e159:0001) > > Subsystem: Unknown device b100:0003 > > > > One is class 0780, one is class 0280. I don't know if this normal, > > but it might be an indication of the problem. > > > > I managed to probe it with zaptel 1.0.8 correctly once after which > > the box paniced. > > What is "it" exactly? IIRC zaptel 1.0.9 is basically 1.0.8 with the > added support for TDM REV I. The TDM400P REV I card. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] TDM400P REV I issues - ProSLIC vs TDM400P
The REV I card shows up in the PCI table as:
02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or
02:05.0 Class 0280: e159:0001)
Subsystem: Unknown device b119:0001
But the REV E/F shows up as:
02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface (or
02:0d.0 Class 0780: e159:0001)
Subsystem: Unknown device b100:0003
One is class 0780, one is class 0280. I don't know if this normal,
but it might be an indication of the problem.
I managed to probe it with zaptel 1.0.8 correctly once after which
the box paniced.
$ sudo /sbin/modprobe wcfxs zaptel
kernel: Zapata Telephony Interface Registered on major 196
kernel: Freshmaker version: 73
kernel: Freshmaker passed register test
kernel: Module 0: Installed -- AUTO FXS/DPO
kernel: Module 1: Installed -- AUTO FXO (FCC mode)
kernel: Module 2: Installed -- AUTO FXO (FCC mode)
kernel: Module 3: Installed -- AUTO FXO (FCC mode)
kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
kernel: Registered tone zone 0 (United States / North America)
One restart later, it fails:
kernel: Zapata Telephony Interface Registered on major 196
/lib/modules/2.4.21-4.EL/misc/wcfxs.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including
invalid IO or IRQ parameters.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.21-4.EL/misc/wcfxs.o: insmod
/lib/modules/2.4.21-4.EL/misc/wcfxs.o failed
/lib/modules/2.4.21-4.EL/misc/wcfxs.o: insmod wcfxs failed
Very confusing. Adding some printk()s to wcfxs.c, for example in
wcfxs_init() just before the call to pci_module_init() and as the
first command in wcfxs_init_one(), shows that pci_module_init()
gets called but wcfxs_init_one() never gets called.
In the PCI table of wcfxs_pci_tbl, if I add
{ 0xe159, 0x0001, 0xb119, PCI_ANY_ID, 0, 0, (unsigned long) &wcfxsi },
(as done revision 1.116 of wctdm.c)
to it, the modprobe works (at least wcfxs_init_one() gets called):
kernel: Freshmaker version: 73
kernel: Freshmaker passed register test
But then, the full story is:
kernel: Zapata Telephony Interface Registered on major 196
kernel: Freshmaker version: 73
kernel: Freshmaker passed register test
kernel: Module 0: Installed -- AUTO FXS/DPO
kernel: Module 1: Installed -- AUTO FXO (FCC mode)
kernel: Module 2: Installed -- AUTO FXO (FCC mode)
kernel: Module 3: Installed -- AUTO FXO (FCC mode)
kernel: Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
and the next attempt gives me:
kernel: Freshmaker version: 73
kernel: Freshmaker passed register test
kernel: ProSLIC on module 0, product 0, version 2
kernel: ProSLIC on module 0 seems sane.
kernel: ProSLIC on module 0 powered up to -74 volts (c6) in 10 ms
kernel: Loop current set to 20mA!
kernel: Post-leakage voltage: 48 volts
kernel: ProSLIC on module 0 powered up to -75 volts (ca) in 0 ms
kernel: Loop current set to 20mA!
kernel: Calibration Vector Regs 98 - 107:
kernel: 98: 11
kernel: 99: 10
kernel: 100: 00
kernel: 101: 00
kernel: 102: 06
kernel: 103: 34
Does this still make sense to anybody?
Edwin
--
Edwin Groothuis |Personal website: http://www.mavetju.org
[EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/
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[Asterisk-Users] Caller ID, Called ID and Forwarded ID
Last month I saw something funny which I can't reproduce anymore: A 0500 number in .au is a service phone number and are forwarded on exchange level to a real phonenumber. So if A calls B it gets forwarded to C. Very simple. Now the funny thing, on the phone of C, I saw both A and B as the "caller id". I've been asking around and trying to get it again with a private 0500 number, but haven't been able to recreate the behaviour. It could be a special Q.931 option, it could be a random fluke of the system, it just could be standard practise. Has anybody ever seen this behaviour or know more of it? Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: can't make my PRI dial out
On Fri, Apr 22, 2005 at 11:10:43AM -0500, [EMAIL PROTECTED] wrote: > I have a full PRI installed on my * machine. I can get inbound calls > just fine but can't make outbound ones. If you run "pri debug span x", you might see this behaviour: PRI debugging with the inbound numbers show that there is a minor difference in the SETUP frame: In the Channel ID, Telstra hasn't set the exclusive bit, AAPT has. Solution for me was in zapata.conf: - spanmap => 1,1,1 + spanmap => 1,1 Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: ISDN BRI + echo cancelling + Fax
On Thu, Apr 14, 2005 at 06:04:20AM -0500, [EMAIL PROTECTED] wrote: > I'm using bristuff on a box with one BRI line. I have echo cancel on. > > However incoming faxes (which routed to a Sipura 2000) fail after about 2 > pages and the fax will report a communication error. > > I switched off echo cancel but ofcourse now I have an extremely annoying > echo. > > Is there anyway to differentiate and switch on/off echo cancel depending > on the type of communication? Adding another bri line just because of this > is not really an option. You might find this one handy: http://bugs.digium.com/bug_view_page.php?bug_id=0003881 [patch] Add E option to the Dial command to explicitely disable echo cancellation Still have to make it available for HEAD. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] Asterisk security problem: authorized SIP users can fake any callerid!
On Fri, Mar 11, 2005 at 01:13:25PM -0600, [EMAIL PROTECTED] wrote: > all that started by investigating what happens if SIP clients are > calling anonymously. > The problem: Every client who is registered as a regular user with > username and secret can fake any callerid in subsequent INVITEs. > Asterisk does not apply an accountcode or callerid from sip.conf. Those > calls end up unbilled and untraceable. I have had this problem too, and was honestly expecting the regexten to show up there instead of the number provided by the user (at least with CALLERIDNUM) > Is there any way to fix this problem - did I misunderstand something, > what am I doing wrong? Besides setting it in the sip.conf (callerid="Foo Bar" <911>), no. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] Digium E1/T1 card with mgetty+sendfax
On Mon, Feb 28, 2005 at 04:33:05AM -0600, [EMAIL PROTECTED] wrote: > On Mon, 28 Feb 2005, Edwin Groothuis wrote: > > > For the project I've used the Eicon DIVA card. It has 8 BRI ports, > > and for about 25% of the time there are 7 or 8 in use. So we want > > to replace it with an E1 card. Only issue is, replace it with what? > > > > The idea we have been playing with was to get a Digium E1 card (we > > already have bought lot of Quad E1 cards :-) and then just put it > > back to back against Asterisk server. And instead of letting > > mgetty+sendfax talk to /dev/ttyI[0-7], we use /dev/zap/[0-30]. > > sendfax (and mgetty) requires a modem interface. The zaptel interfaces are > raw tdm interfaces. SpanDSP could be made to provide a smartmodem > interface but no such code exists yet (as far as I know). Aha, that makes sense. On a different note, isdn4linux provides such an interface. Only then I need a way have i4l talking with the zaptel driver. Wish I understood more of this stuff... :-/ Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] Digium E1/T1 card with mgetty+sendfax
Hi, For the project I've used the Eicon DIVA card. It has 8 BRI ports, and for about 25% of the time there are 7 or 8 in use. So we want to replace it with an E1 card. Only issue is, replace it with what? The idea we have been playing with was to get a Digium E1 card (we already have bought lot of Quad E1 cards :-) and then just put it back to back against Asterisk server. And instead of letting mgetty+sendfax talk to /dev/ttyI[0-7], we use /dev/zap/[0-30]. Has anybody else ever tried this? Success- and horror stories are welcome! Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] /var/run/asterisk.ctl configuration
People who use "asterisk -r" already know that if you don't try it as root (for example webservers), you don't get access. Of course you can chmod it aech time you start asterisk, but that gets annoying after two times. I have made a patch (against HEAD and 1.0.5) which makes it possible for you (=asterisk admin) to configure the ownership and permissions of /var/run/asterisk.ctl. If you need this feature, see http://bugs.digium.com/bug_view_page.php?bug_id=0003558 for the patches. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] extensions.conf - redundancy removal
The number 9335 3510 is sent by the telco.
The number 1414 02 9335 3510 is sent by an PABX.
The number 02 9335 3510 is the one which is the real telephone number:
exten => 93353510,1,Macro(call-local-sip,edwin,02${EXTEN})
exten => 0293353510,1,Macro(call-local-sip,edwin,${EXTEN})
exten => 14140293353510,1,Macro(call-local-sip,edwin,${EXTEN:4})
Is there a way to do this in one line?
--
Edwin Groothuis |Personal website: http://www.mavetju.org
[EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/
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Re: [Asterisk-Users] IPv6 and Asterisk?
On Wed, Nov 24, 2004 at 04:48:04AM -0600, [EMAIL PROTECTED] wrote: > > Can we start backwards and write coding guidelines, like > > "don't use strings like ip:port with a colon, since a colon is a > > significant > > character in IPv6 addresses" Anybody who is doing IP addresses while talking about IPv6 will quickly find out why DNS was invented. I'm glad to test IPv6 internally, but my IPv6 tunnel (via FreeNet6) is too slow for VoIP. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] Error "WARNING[-150101888]" when starting Asterisk.
On Sun, Nov 21, 2004 at 04:34:38PM -0600, [EMAIL PROTECTED] wrote: > [chan_zap.so]Nov 21 20:37:05 WARNING[-150101888]: loader.c:248 The warning-number your quoted, -150101888, is actually the process id of the thread the error gets reported about. See also http://bugs.digium.com/bug_view_page.php?bug_id=0002745 Not that it helps you in this case, just explaining a little bit. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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
Config files (was: Re: [Asterisk-Users] Interrupting MusicOnHold while call in queue ?)
On Fri, Nov 19, 2004 at 07:34:19PM +1100, Edwin Groothuis wrote: > > > You can either do a "make samples" (which will overwrite your existing > > configuration) or look in the "configs" subdirectory in the asterisk > > source tree for queues.conf.sample. > > It would be nicer if "make samples" would install the configs as > .sample, for example: /etc/asterisk/iax.conf.sample. > > That way it's safe to do "make samples" (which is always a good > thing) and it's easy to add changes because all you have to do is > add these *.sample files to CVS and do a "cvs diff -u" after you've > installed a new version. That way new lines and changes in defaults > are spotted and can be put in the live configuration files. http://bugs.digium.com/bug_view_page.php?bug_id=0002908 -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] Interrupting MusicOnHold while call in queue ?
> You can either do a "make samples" (which will overwrite your existing > configuration) or look in the "configs" subdirectory in the asterisk > source tree for queues.conf.sample. It would be nicer if "make samples" would install the configs as .sample, for example: /etc/asterisk/iax.conf.sample. That way it's safe to do "make samples" (which is always a good thing) and it's easy to add changes because all you have to do is add these *.sample files to CVS and do a "cvs diff -u" after you've installed a new version. That way new lines and changes in defaults are spotted and can be put in the live configuration files. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] TE410P - How many can I have?
On Thu, Nov 18, 2004 at 10:56:26AM -0600, [EMAIL PROTECTED] wrote: >9. TE410P - How many can I have? (Matthew Boehm) > > I have a Dell Poweredge 6450, 4 proc Xenon with 1Gb ram and the following > PCI abilities: > > Bus type . . . . .. . three peer PCI buses: two 64-bit buses and one 32-bit > bus > Expansion slots .. seven hot-pluggable PCI slots > two 64-bit/66 MHz, > four 64-bit/33 MHz, > one 32-bit/33 MHz http://www.digium.com/index.php?menu=whatpcislot has a nice overview of the layout. You need the bus which looks like the one marked with a 2: It has the nodge near the chassis. Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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] FreeBSD Asterisk and G729 codec
On Thu, Nov 18, 2004 at 03:27:40AM -0600, [EMAIL PROTECTED] wrote: > I just purchased 10 G729 licenses for my asterisk box from Digium I was able > to register the key. But when i start asterisk it fails with the error > message: > [codec_g729a.so]Nov 18 09:27:01 WARNING[135073792]: loader.c:248 > ast_load_resource: Shared object "libc.so.6" not found > Nov 18 09:27:01 WARNING[135073792]: loader.c:380 load_modules: > Loading module codec_g729a.so failed! Isn't the codec_g729a.so not a binary you download instead of build yourself? In that case you can't run it with the FreeBSD version of Asterisk. You need to compile it on a linux machine and copy the libraries and binaries. Of course it would be much better if Digium would provide a FreeBSD binary of the library :-) Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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 (or IAX) modem driver
Hello, This might be a wild chase, but I'm looking for a driver / daemon / asterisk app which on one side speaks SIP or IAX, and on the other side is a modem emulator. Or to explain with an example: On my *nix box I want to start a terminal program (minicom, kermit, whatever), connect to a virtual modem which then connects to a SIP server. And when a call comes in on that SIP server, it sets up a connection to the modem program which sends an RING to the terminal program. Does anybody know if such a driver / daemon / asterisk app exists (for Linux preferably). Edwin -- Edwin Groothuis |Personal website: http://www.mavetju.org [EMAIL PROTECTED]| Weblog: http://weblog.barnet.com.au/edwin/ ___ 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
