Displaying 4 results from an estimated 4 matches for "dialed_number".
2015 Aug 14
2
chan_sip.c: Retransmission timeout reached on transmission
...th DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 111") in new stack
[Aug 12 19:21:14] VERBOSE[17115] pbx.c: -- Executing
[s at macro-dialout-trunk:21] Goto("SIP/143-000001d8", "s-CHANUNAVAIL,1") in
new stack
*[3]*
Retransmitting #3 (no NAT) to PROVIDER-IP:5060:
INVITE sip:dialed_number at PROVIDER-IP SIP/2.0
Via: SIP/2.0/UDP PBX-PUBLIC_IP:5060;branch=z9hG4bK06c2c701
Max-Forwards: 70
From: "PBX-DID" <sip:outbound-trunk at PROVIDER-IP>;tag=as27ef83ae
To: <sip:dialed_number at PROVIDER-IP>
Contact: <sip:outbound-trunk at PBX-PUBLIC_IP:5060>
Call-ID: 6b9ad...
2005 May 13
2
In/out calls from/to same sip provider
...g calls regarding one provider, but not
both.
I've also tried the sample sintax
"exten =>_42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)"
that comes with the distribution (debian-sarge), but only to get asterisk
unable to create sip channel because
"host dialed_number@real_sip_server_address doesn't exist". The address
is that of the provider.
voip.org and asteriskdocs.org seems to lead me nowhere.
I must be missing something obvious, but can't figure out what it is.
Anybody?
Thanks.
--
Pizco Dominguez
-----------------------------------------...
2004 Apr 13
2
T100P E&M Wink Trunk
...&M
so we could pass an unlimited number of DIDs to the trunk as apposed to
FXS loopstart signaling. I can make outbound calls no problem, but I am
having problems with the dial plan for inbound calls. The way they setup
the trunk inbound calls have a dialed number as
"*<callerid>*<dialed_number>". I do not know how to parse this out and
map it in the dial plan. Are there substr functions I can use? Can I
just call SetCIDNum on an INBOUND call to get the callerid functions
working?
Here is what I see in the log when a call comes in:
-- Starting simple switch on 'Zap/24-1&...
2006 Dec 10
1
Problem faxing with SPA2100 in passthru mode.
...PA2100's logs, but I can't see anything of
interest (and I couldn't find any documentation about this logs at
Sipura's website). The ATA seems to dial correctly but, after a few
seconds, it hangs the call (CC:Failed w/ Calling)
[0]Off Hook
2. Report digit first_digit_of_dialed_number (1)(40 ms)
2. Report digit second_digit_of_dialed_number (1)(40 ms)
.... etc ...
2. Report digit last_digit_of_dialed_number (1)(40 ms)
Calling:dialed_number@asterisk_box_ip_address:0
[0:0]AUD ALLOC CALL (port=16434)
[0:0]RTP Rx Up
[0:0]ENC INIT 8...