[asterisk-users] Responding to SIP OPTIONS
I understand that it is customary for SIP User Agents to send OPTIONS packets every now and then to check that a peer is still alive and well. Indeed I understand that Asterisk itself sends them if qualify is set to yes in the peer configuration. How is one supposed to configure the dialplan so that Asterisk responds correctly to these requests? At the moment, I'm seeing Looking for s in default and then a 404 Not Found being returned - which can't be right. Thanks! Alex Lake DIGITAL MAIL LIMITED ___ --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] Problem with Asterisk 1.4.0-beta3 and Digium TE405P
Hope this is the right place to report/ask for help...
Have have a 1.2.7.1 installation running reasonably happily for a while.
Thought we might give 1.4.0b3 a go. Ran it on a local test machine (that
has the single port card) and all was well. However, when I run it on a
machine with the 4-port card, it has trouble with Zaptel. I attach some
diagnostics and welcome any suggestions (including requests for more
information). We're running this on a Del SC430 with Ubuntu 5.10.
Thanks!
Alex
-
[EMAIL PROTECTED]:/etc/modprobe.d# /etc/init.d/asterisk start
Starting Asterisk PBX: Notice: Configuration file is /etc/zaptel.conf
line 22: Unable to read Zaptel version information.
Zaptel Version: ôoô·´vô·ÿwHVä¿Èó·Vä¿`vô·
Echo Canceller:
Configuration
==
SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 2: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 3: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
SPAN 4: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)
124 channels configured.
ZT_CHANCONFIG failed on channel 1: Inappropriate ioctl for device (25)
[EMAIL PROTECTED]:/etc/modprobe.d# strace ztcfg
execve(/sbin/ztcfg, [ztcfg], [/* 23 vars */]) = 0
uname({sys=Linux, node=m900a, ...}) = 0
brk(0) = 0x80a2000
access(/etc/ld.so.nohwcap, F_OK) = -1 ENOENT (No such file or
directory)
old_mmap(NULL, 8192, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xb7fa6000
access(/etc/ld.so.preload, R_OK) = -1 ENOENT (No such file or
directory)
open(/etc/ld.so.cache, O_RDONLY) = 3
fstat64(3, {st_mode=S_IFREG|0644, st_size=40467, ...}) = 0
old_mmap(NULL, 40467, PROT_READ, MAP_PRIVATE, 3, 0) = 0xb7f9c000
close(3)= 0
access(/etc/ld.so.nohwcap, F_OK) = -1 ENOENT (No such file or
directory)
open(/lib/tls/i686/cmov/libm.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0`3\0\000...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0644, st_size=136976, ...}) = 0
old_mmap(NULL, 139424, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE,
3, 0) = 0xb7f79000
old_mmap(0xb7f9a000, 8192, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x2) = 0xb7f9a000
close(3)= 0
access(/etc/ld.so.nohwcap, F_OK) = -1 ENOENT (No such file or
directory)
open(/lib/tls/i686/cmov/libc.so.6, O_RDONLY) = 3
read(3, \177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\260O\1...,
512) = 512
fstat64(3, {st_mode=S_IFREG|0644, st_size=1229936, ...}) = 0
old_mmap(NULL, 1236124, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE,
3, 0) = 0xb7e4b000
old_mmap(0xb7f73000, 16384, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 3, 0x128000) = 0xb7f73000
old_mmap(0xb7f77000, 7324, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0xb7f77000
close(3)= 0
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xb7e4a000
old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS,
-1, 0) = 0xb7e49000
mprotect(0xb7f73000, 4096, PROT_READ) = 0
set_thread_area({entry_number:-1 - 6, base_addr:0xb7e496c0,
limit:1048575, seg_32bit:1, contents:0, read_exec_only:0,
limit_in_pages:1, seg_not_present:0, useable
:1}) = 0
munmap(0xb7f9c000, 40467) = 0
open(/dev/zap/ctl, O_RDWR)= 3
brk(0) = 0x80a2000
brk(0x80c3000) = 0x80c3000
open(/etc/zaptel.conf, O_RDONLY) = 4
fstat64(4, {st_mode=S_IFREG|0744, st_size=256, ...}) = 0
mmap2(NULL, 131072, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1,
0) = 0xb7e29000
read(4, span=1,1,0,ccs,hdb3\nbchan=1-15\nd..., 131072) = 256
read(4, , 131072) = 0
close(4)= 0
munmap(0xb7e29000, 131072) = 0
ioctl(3, 0x40244a12, 0x80a0a60) = 0
ioctl(3, 0x40244a12, 0x80a0a84) = 0
ioctl(3, 0x40244a12, 0x80a0aa8) = 0
ioctl(3, 0x40244a12, 0x80a0acc) = 0
ioctl(3, 0x80844a05, 0xbfcbaaec)= 0
ioctl(3, 0x404c4a13, 0x808daac) = -1 ENOTTY (Inappropriate ioctl
for device)
write(2, ZT_CHANCONFIG failed on channel ..., 71ZT_CHANCONFIG failed
on channel 1: Inappropriate ioctl for device (25)
) = 71
close(3)= 0
exit_group(1) = ?
---
/etc/zaptel.conf
span=1,1,0,ccs,hdb3
bchan=1-15
dchan=16
bchan=17-31
span=2,2,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
span=3,0,0,ccs,hdb3,crc4
bchan=63-77
dchan=78
bchan=79-93
span=4,0,0,ccs,hdb3
bchan=94-108
dchan=109
bchan=110-124
loadzone=uk
defaultzone=uk
___
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[asterisk-users] Asterisk and failover
I'm trying to implement a dual Asterisk Box setup where our carrier will be delivering SIP traffic to one of 2 units. The idea is that to have a high uptime by handling traffic on box A while box B is being upgraded, then box B takes over the traffic, allowing box A to be upgraded. The question is how can we drive Asterisk to make this happen? I need some primitives like stop accepting calls and start accepting calls. One way (I guess) is to have a dial plan that just responds to any INVITE with a 3xx message and switch that in with a file rename and a console reload operation. Another way would involve the use of the OPTIONS feature of SIP, but I can't see where this is configured. Any ideas or experience on this matter? Alex ___ --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] Asterisk and failover
It's beginning to look as though Asterisk can't send 302 responses. Is this really the case? I'm trying to implement a dual Asterisk Box setup where our carrier will be delivering SIP traffic to one of 2 units. The idea is that to have a high uptime by handling traffic on box A while box B is being upgraded, then box B takes over the traffic, allowing box A to be upgraded. The question is how can we drive Asterisk to make this happen? I need some primitives like stop accepting calls and start accepting calls. One way (I guess) is to have a dial plan that just responds to any INVITE with a 3xx message and switch that in with a file rename and a console reload operation. Another way would involve the use of the OPTIONS feature of SIP, but I can't see where this is configured. Any ideas or experience on this matter? Alex ___ --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 ___ --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] Asterisk forwarding SIP with Remote-Party-ID
I'm finding that I'm a bit disappointed that Asterisk doesn't naturally forward the Remote-Party-ID from inbound SIP calls (where trustedrpid=yes) to outbound SIP calls. I guess this is going to be something we have to use SER for, unless we make our own custom build (which I'm reluctant to do). Is there a good reason that the main version doesn't do it? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] Asterisk and NAT
You've not said much about your firewall setup. I presume you've opened up 5060 and RTP ports? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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: AW: [Asterisk-Users] Asterisk and NAT
I guess you could post your config files here and hope that someone feels inclined to look them over! ;-) ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] SIP make outside call
I'm going to dip my toe in the water to help out here, although I'm just a newbie... It looks to me as though your x-lite is coming in and being assigned the sip context, which includes just the right to call internal destinations. (Bizarrely?) Your default context seems to allow everything (and double-includes internal!) You could change extensions.conf to this, but it's still somewhat odd... [sip] include = internal include = outgoing [default] include = internal include = outgoing Best of luck, Alex ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] One-way audio with VPN
Thanks for that. We did track it down to a problem with native bridging. In this case, Asterisk assumed that the VPN was publicly accessible - but it isn't! The fix we've found is to setup all VPN-based sip devices with canreinvite=no, but I'm not sure if this is the best way to do that. ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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] One-way audio with VPN
I've got a one-way audio problem, but I've looked through a few documents on the subject and I'm not sure that it's the same issue. User A calls a local Asterisk user B via a public SIP gateway (voiptalk.org) using (sip:[EMAIL PROTECTED]) B is connected to the Asterisk server via VPN B is registered (and has successful bi-directional conversations with other users on the VPN) Asterisk correctly forwards the call via SIP and B's phone rings and is answered, but B can't hear A So there appears to be an audio-path blockage from A via Asterisk to B. Now if A leaves a voicemail message on the asterisk box, that's fine (the sound file contains a recording of A's voice!) Therefore, it looks like the problem is to do with the forwarding of RTP packets by Asterisk from A (Internet origin) to B (VPN). Any ideas? ___ --Bandwidth and Colocation sponsored by Easynews.com -- 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
