Re: [Asterisk-Users] app_sms: problems sending a sms
Hello Seshu,
i think you solved your problem in the meantime, but here
are my points (for archive purposes), after it works here now.
> Thanks Steffen. Please update me if this ever works.
The problem was (probably), that i put some lines in
extension.conf for sending a sms, and triggered
this by calling this extension. Probably the sms app
can't get the right channel then.
After changing this, i had in the extensions.conf:
[smsdial]
exten => _X.,1,SMS(default,,${EXTEN},${MSG})
exten => _X.,2,SMS(default)
exten => _X.,3,Hangup
exten => h,1,Hangup
(like a poster here posted it already) and put a call file
in /var/spool/asterisk/outgoing/
[EMAIL PROTECTED]:/tmp# cat testsms
Channel: Zap/g1/0090032669000
CallerID: SMS <35910>
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: smsdial
Extension: 0179XXX
Priority: 1
SetVar: MSG=Text to send
That works here now.
A few hints:
- If it doesn't work, try to call the SMSC with your phone. You should
hear the sound of a bird chirp and the SMSC should hangup after a
few seconds.
If you get something like "This number is not complete", try to add
a zero at the end of the number and repeat.
If nothing happens, check out if calling the number of the SMSC is
allowed from your line.
- Put the number of your desk phone in your call file instead of the
number of your SMSC. After putting the call file in the outgoing dir,
your desk phone should ring. Now you know that your asterisk-setup
is ok (or not).
- Get and read the ETSI ES 201 912 standard to understand what's going
on and to know the meaning of the message type codes the sms app
is printing. I think this is a good idea to put this in the wiki.
HTH,
Steffen
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Re: [Asterisk-Users] chan_cornet
Hello, > Am i right when i suppose that the chan_cornet will replace the oh.323. > > [OPTIPOINT400_HFA]--[HIPAT4K][chan_cornet][ASTERISK]--[OPTIPOINT400_SIP]. Yes, that's the idea. Let's look if we have success :) cu, Steffen ___ 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] chan_cornet
Hello, > but what do you mean with *new hipath version doesn't > support H.323 anymore*? What version are you talking about? > As far as i know the new version of HiPath4000 V2.0 still > supports H.323 (STMI2). HiPath 3000 -> H.323 Support HiPath 4000 -> NO H.323 Support and nothing else but cornet (voip). cornet-ip is basic h.323 + addons, but it does not work with standard h.323 devices. cu, Steffen ___ 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] chan_cornet
Hello, > I dont know if Steffen's chan_cornet is working. I emailed him, but with no > result. You are not patient enough ;) You got an answer one minute ago ;) No it is not ready, it is work in progress. At the moment i'm forced to get a Asterisk->SMS Gateway working here, with our old PBX, but i hope it works soon so that i can proceed with chan_cornet. At the moment, Optipoint 400 and 600s can register to the chan_cornet, and one can call them so that they ring. There is some little work to be done, to get the voice working (a bit H.323 stuff), and with a little editor (for entering the numbers, the phone can't handle this), the Phone2PBX part should work. And then the next goal is the PBX2PBX stuff. That newer HiPath PBXs are worse, coz Siemens dropped the H.323-Support, that means they do not support one standard voip protocol. They said that they will support SIP in the future, but they say this for more than a year now. That means, one can connect now that PBXs with TDM-Lines (S2M, BRI) or to another cornet-ip supporting device like IPDAs (smaller siemens pbxs that connect to a main PBX) or PBXs and hopefully, chan_cornet sometime ;). cu, Steffen ___ 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] app_sms: problems sending a sms
Hello Seshu, no it still does not work. I started to debug this thing, and as far as i can say, the problem is that app_sms does not recognize the initial connection established packet, that the SM-SC sends when answering the call. I don't know now what exactly the problem is at the moment, possible that it is the level of the signal. I will examine this in the next days. cu, Steffen ___ 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] app_sms: problems sending a sms
Hello,
i try to send out a sms, but with no success.
The trunk is a E100P, and the sms should go out to the
Telekom SM-SC. What i want to to at the first run is,
sending out a sms when a certain number is dialed.
I tried:
In extensions.conf:
exten => 35953,1,SMS(${TRUNK}/9350193010,,0179NUMBER,"Hi there")
exten => 35953,2,SMS(${TRUNK}/9350193010)
exten => 35953,3,Hangup
exten => 35954,1,Dial(${TRUNK}/9350193010)
and get:
tkserv*CLI>
-- Executing Goto("SIP/35903-da57", "voiplocal|35953|1") in new stack
-- Goto (voiplocal,35953,1)
-- Executing SMS("SIP/35903-da57", "Zap/g1/9350193010||0179NUMBER|"Hi
there"") in new stack
-- Executing SMS("SIP/35903-da57", "Zap/g1/9350193010") in new stack
-- SMS TX 92 01 FF 6E 00 00...
-- Executing Hangup("SIP/35903-da57", "") in new stack
== Spawn extension (voiplocal, 35953, 3) exited non-zero on 'SIP/35903-da57'
935 is the prefix to go out to the world via a telekom PRI line.
Sometimes i hear a chirp like the sound of a bird, sometimes i
get this "SMS TX 92 01 FF 6E 00 00..." line, sometimes nothing
happens but a hangup after a few seconds.
(0179NUMBER is the number of the cell-phone).
When i call the 35954 via a SIP Phone, i hear always one chirp,
and a hangup after a few seconds, so i guess the call reaches
the SM-SC.
Does someone know whats wrong?
cu,
Steffen
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Re: [Asterisk-Users] Nortel Phones.
Hi Carl, > The source code is available at : > http://www.mlkj.net/UNISTIM/voi.tar.bz2 i had a quick look at you code. Nice piece of work :) But what's the purpose of your code? Seems much too big for me for a simple proof of concept (POC). Do you want to expand this code in future? Or is it made to implement it in other (software) PBXes? This is very nice to see for me, because i reverse-engineered the Siemens CorNet-IP proprietary protocol (used in HiPath PBXes) and try to write a chan_cornet at the moment. This are simple phones that act like a terminal (or thin client), the whole intelligence is in the server. The nice thing with this system phones is - compared to standard protocols like H.323 or SIP - that they support a lot of features that the standard protocols do not. You do not want to write a chan_unistim? ;) cu, Steffen ___ 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] Optipoint 400 Standard Sip
Hello,
> I tried nearly everything, up and downgraded firmware, but nothing worked.
> I phoned with a siemens engineer, and he told me, that this version is not
> 100% sip confirm.
> But there will be a workaround.
if you mean the not-registering problem (and it works when the secret
in sip.conf is commented out), try the patch below.
The patch worked with asterisk-0.7.2, and several Optipoint 400 SIP
(2.3.12) work well with asterisk here.
Credits go out to Wilhelm Wimmreuter who posted a patch for this
problem at 30 May 2003 at this ML. Parts of his patch found their
way in the asterisk source tree, the patch included here is what's
remaining to get this nice but expensive siemens phones working.
cu,
Steffen
diff -Naur asterisk-cvs-20031029/channels/chan_sip.c
asterisk-cvs-20031029-new/channels/chan_sip.c
--- asterisk-cvs-20031029/channels/chan_sip.c 2003-10-25 19:41:02.0 +0200
+++ asterisk-cvs-20031029-new/channels/chan_sip.c 2003-10-29 21:35:00.0
+0100
@@ -2219,7 +2219,10 @@
snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact,
p->expiry);
snprintf(tmp, sizeof(tmp), "%d", p->expiry);
add_header(resp, "Expires", tmp);
- add_header(resp, "Contact", contact);
+/*ww lwc change header to copy
+add_header(resp, "Contact", contact);
+*/
+copy_header(resp, req, "Contact");
} else {
add_header(resp, "Contact", p->our_contact);
}
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Re: [Asterisk-Users] Re: E100P driver overwrites memory used bye linux-kernel
Hello, yes, you are right, the MMX turned on was the cause of the problems. Without, it runs without any problems. But i don't think that the Athlon XP doesn't like MMX. Do you know what fails if MMX is used with this CPU? cu, Steffen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: E100P driver overwrites memory used bye linux-kernel
Hi, i did some tests. If the kernel modules are loaded Module Size Used byNot tainted wct1xxp11488 31 zaptel176256 64 [wct1xxp] ppp_generic19068 0 [zaptel] slhc5392 0 [ppp_generic] and the E1 is connected to the PBX, everything is ok. Then i started asterisk, it opens the D-Channel and everything is still ok. I left the system in this state and it survived one night without problems. But immediately after the first call (B-Channel) the system memory is overwritten and bad things happen. I looked through the wct1xxp.c and it looks like the D-Channel (HDLC) is handled between hardware and driver like the B-Channels (no HDLC decoding in hardware), right? That would mean that it is probably software problem. Is there a way to do some test calls without asterisk? cu, Steffen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E100P driver overwrites memory used bye linux-kernel
Hi, we have a Digium E100P in use with asterisk and the driver of the E100P overwrites important memory locations of the kernel. The side-effects are malloc-errors when simple shell commands are used, unable to compile anything (internal compiler errors), files that are edited are in the state as before editing. The memory is ok, tested this with memtest86 and no problems occur if the driver is not loaded. System: - CPU: Athlon XP 2600+ - RAM: 1 GB DDR400 RAM - BRD: EPOX 8KRA2i KT600 Kernel 2.4.21 (4GB) High Memory Support [*] HIGHMEM I/O support [*] MTRR (Memory Type Range Register) support [*] Local APIC support on uniprocessors [*] IO-APIC support on uniprocessors can the highmem i/o support be the cause of the problems? Has anyone had this problem, too? cu, Steffen Koepf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
