Displaying 7 results from an estimated 7 matches for "11b2".
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2006 Jan 26
3
Chan_capi on builds 7955>8320 strangeness
Hello All,
I am having an odd problem with Armin's chan-capi_cm on builds higher
than 7955.
It would seem that this happens on anything higher than 7955.
What is happening is the isdn is ringing, then asterisk does a goto-if
and just hangs.
Asterisk itself is ok, but the isdn then rings out or busys out on the
other side.
Outgoing works fine, this only seems to effect incoming.
I
2004 Apr 01
1
I didn't want to bother the list with this, but...
...namic
dtmfmode=inband
username=gjunker
auth=md5
secret=xxxxxxxxxxxxxxxxxxxxxxxxxxx ; generated per instructions in the
Wiki
Asterisk sip debug output:
Sip read:
REGISTER sip:voip.shockwaveaudio.com SIP/2.0
Content-Length: 0
Contact: <sip:gjunker@192.168.1.236:5060>
Call-ID: ED30CAFA-1DD1-11B2-B310-C0D342A71FF4@192.168.1.236
From: <sip:gjunker@voip.shockwaveaudio.com>;tag=2798956518
CSeq: 15 REGISTER
To: <sip:gjunker@voip.shockwaveaudio.com>
Via: SIP/2.0/UDP 192.168.1.236:5060...
2005 Feb 09
1
calling problem in cvs verison on fedora core2
...n default
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.0.147;branch=z9hG4bKc0a800930131c9b1420a015c4a64c5410000000a
From:
"root"<sip:2000@192.168.0.203>;tag=4145368953696567536
To: <sip:3000@192.168.0.203>;tag=as3521c065
Call-ID:
C3F18200-1DD1-11B2-A327-F7B2F6A2C3A6@192.168.0.147
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:3000@192.168.0.203>
Content-Length: 0...
2005 Feb 10
4
why asterisk is replying 404 Not Found
...n default
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.0.109;branch=z9hG4bKc0a8006d0131c9b14209a71a2c53e29900000004
From:
"root"<sip:3000@192.168.0.203>;tag=4122263641472432750
To: <sip:2000@192.168.0.203>;tag=as2f7c300d
Call-ID:
F81AB0F0-1DD1-11B2-91B7-F323B891C30C@192.168.0.109
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2000@192.168.0.203>
Content-Length: 0
__________________________________
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http://promo...
2010 Apr 30
2
Continuing after a TIMEOUT(absolute)
...n new stack
== Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
70.124.61.17-082a69a8' in macro 'phonenum'
== Spawn extension (macro-phonenum, s, 7) exited non-zero on 'SIP/
70.124.61.17-082a69a8'
Scheduling destruction of SIP dialog 'D8FE9724-1DD1-11B2-9F1A-
A4EF9DB84584 at 192.168.1.98' in 32000 ms (Method: ACK)
set_destination: Parsing <sip:70.124.61.17:5060> for address/port to
send to
set_destination: set destination to 70.124.61.17, port 5060
Reliably Transmitting (NAT) to 70.124.61.17:5060:
BYE sip:70.124.61.17:5060 SIP/2.0
<...
2005 Mar 23
6
Problem parsing unusual SIP/SDP
...ed:
SIP/2.0 183 Session Progress
v:SIP/2.0/UDP [my public IP]:5060;branch=z9hG4bK62d91cea
CSeq:103 INVITE
i:7a1791cf52d6f3dc2d12b208051d0a21@[provider].com
f:"Test User" <sip:[my phone number]@[provider].com>;tag=as341d210b
t:<sip:18003253535@[provider].com>;tag=b6e96dae-1dd1-11b2-a01e-b03162323164+b6e9
6dae
m:sip:18003253535@192.168.201.25:5075
c:application/sdp
l:170
v=0
o=- 3459442714 3459442714 IN IP4 192.168.201.25
s=SIP Call
c=IN IP4 192.168.201.11
t=0 0
m=audio 52322 RTP/AVP 0
c=IN IP4 [provider public IP]
a=rtpmap:0 PCMU/8000
The "200 OK" arrives with sim...
2005 Mar 16
1
Error in placing call file in directory
Greetings *`s,
I have created a call file and am manually placing it in
/var/spool/asterisk/outgoing, but I get the following errors in the log
file :
===========
Mar 16 15:26:57 DEBUG[2054]: Auto destroying call
'E36A0D74-1DD1-11B2-AC8B-860A22D62D71@192.168.204.10'
Mar 16 15:27:43 WARNING[2054]: Unable to open
/var/spool/asterisk/outgoing/chris.call: Permission denied, deleting
Mar 16 15:27:43 WARNING[2054]: Failed to scan service
'/var/spool/asterisk/outgoing/chris.call'
===============================
I have ch...