Displaying 5 results from an estimated 5 matches for "0xxxxxxxx".
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2006 Jun 12
5
use AT320 international call
Hi all,
The firmware I used is pa168s_iax2_us_151011.bin.
My problem is the handset dial before I finished key in all
the numbers, no matter how fast I managed to press the keys.
It appeared it always dialed immediately, for example "011862",
when I actually ment to dial 0118620xxxxxxxx. Thus left the
remaining numbers "0xxxxxxxx" unsent.
The handset had its dial plan disabled. It configured to use
iax protocol. My extensions.conf has this:
exten=_01186.,1, dial(SIP/<voipprovider>,60)
and it works fine with other iaxy and Cisco ATA.
Anyone encounter this s...
2009 Dec 17
2
Integrate a CPE with Asterisk in MGCP
...e call goes down
after some seconds :
-- Executing Answer("MGCP/aaln/1 at 020202020202-1", "") in new stack
-- MGCP mgcp_answer(MGCP/aaln/1 at 020202020202-1) on aaln/1 at 020202020202-1
-- Executing Dial("MGCP/aaln/1 at 020202020202-1",
"SIP/mgcp-out/0xxxxxxxx") in new stack
-- Called mgcp-out/0xxxxxxxxxx
Dec 17 18:34:47 NOTICE[17889]: chan_mgcp.c:1656 find_subchannel_and_lock:
Gateway '020202020202' (and thus its endpoint 'virtual/nat-timeout') does
not exist
-- SIP/mgcp-out-09d998c0 is making progress passing it to
MGCP/aal...
2006 Jan 18
0
asterisk 1.2 bristuff and sms
...e "call" from the telecom operator, and it starts the SMS
application, but then i get the following error.. Any idea if this is
due to bristuff or my implementation of SMS ? (I used voip-info.org as a
reference)
thanks in advance
log:
VERBOSE[7905] logger.c: -- Goto (custom-smsrx,0XXXXXXXX,1)
VERBOSE[7905] logger.c: -- Executing Verbose("Zap/1-1", "Receiving
SMS from 0171701") in new stack
VERBOSE[7905] logger.c: Receiving SMS from 0171701
VERBOSE[7905] logger.c: -- Executing SMS("Zap/1-1", "default|a") in
new stack
VERBOSE[7905] logg...
2020 Mar 28
0
[klibc:update-dash] Implement stpcpy() and stpncpy()
..., temp, end - temp, strlen(temp));
+ if (end != temp + 3 || memcmp(temp, "123\0XXXXX", 10))
+ goto error;
+
+ clear_temp();
+ end = stpcpy(temp, "");
+ printf("'%s'len:%zu strlen:%zu\n", temp, end - temp, strlen(temp));
+ if (end != temp || memcmp(temp, "\0XXXXXXXX", 10))
+ goto error;
+
+ clear_temp();
+ end = stpcpy(temp, "1234567");
+ printf("'%s'len:%zu strlen:%zu\n", temp, end - temp, strlen(temp));
+ if (end != temp + 7 || memcmp(temp, "1234567\0X", 10))
+ goto error;
+
+ printf("\n");
+ printf(&quo...
2009 Oct 05
3
Questions about app_jack.c
...JACK_HOOK from app_jack.so, sound is sent but no one is received.
My extensions.conf :
exten => _0.,1,Answer
exten =>
_0.,n,Set(JACK_HOOK(manipulate,c(asterisk))i(from_voip:input)o(to_voip:outpu
t)))=on)
exten => _0.,n,Dial(SIP/freephonie-out/${EXTEN:1})
Asterisk command :
console dial 0xxxxxxxx
2) Jackd works well with anothers applications when I force them to use jack
as input/output. -> probably not a jack configuration problem.
3) If I kill jackd and I use chan_alsa.so with the real soundcard, it works.
-> probably not a network or sip configuration problem.
4) If I replace...