Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
Steve Totaro wrote: On Mon, Apr 7, 2008 at 6:36 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 07, 2008 at 10:37:42AM +0100, James Williamson wrote: Tzafrir Cohen wrote: On Mon, Apr 07, 2008 at 06:11:02AM +0100, James Williamson wrote: Snap, Well, after trying to buying a TDM400P and then getting persuaded to buy a TDM410P because they no longer sell the 400 model I'd say I'm not impressed. It took three 2.6 kernel builds (zaptel 1.4 won't even build with the latest kernel release) What version have you tried? Of Zaptel and of the kernel? AFAIK 1.4.9.2 builds with latest kernel (or maybe there's actually a small warning with our drivers, fixed in SVN) Please provide an error log. Yes, zaptel 1.4.9.2 does compile against a 2.6.24.4 source tree, although the latest release on the website (something like a week ago) was 1.4.8 which doesn't: [EMAIL PROTECTED] zaptel-1.4.8]# make make[1]: Entering directory `/usr/local/src/zaptel-1.4.8' make -C /lib/modules/2.6.24/build SUBDIRS=/usr/local/src/zaptel-1.4.8 HOTPLUG_FIRMWARE=yes modules make[2]: Entering directory `/usr/src/linux-2.6.24' WARNING: Symbol version dump /usr/src/linux-2.6.24/Module.symvers is missing; modules will have no dependencies and modversions. scripts/Makefile.build:46: *** CFLAGS was changed in /usr/local/src/zaptel-1.4.8/Makefile. Fix it to use EXTRA_CFLAGS. Stop. make[2]: *** [_module_/usr/local/src/zaptel-1.4.8] Error 2 make[2]: Leaving directory `/usr/src/linux-2.6.24' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/local/src/zaptel-1.4.8' make: *** [all] Error 2 Yeah, fixed long ago in 1.4.9 (even though there's a very simple workaround for it) -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Another vote for 1.2.X. Openvox is the same as Digium TDM400P, it is the reference design and the cards are made very well. I suggest trying a Sangoma card. I've installed the 1.2.x zaptel drivers, this still doesn't work. Is there anyone in the UK who's successfully got a TDM410P to support caller id or am I just wasting my time? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
Tzafrir Cohen wrote:
On Tue, Apr 08, 2008 at 09:42:48AM +0100, James Williamson wrote:
I've installed the 1.2.x zaptel drivers, this still doesn't work. Is
there anyone
in the UK who's successfully got a TDM410P to support caller id or am I just
wasting my time?
Before going further in wasting time, what do you have in zapata.conf ?
(And for the record: I don't think 1.2 should be any better than 1.4 in
picking up caller ID)
I'm running a 2.6.22 kernel, zaptel 1.4.9.2 and asterisk 1.4.14. I've
ensured the driver is loaded in opermode=UK,
dmesg output:
wctdm24xxp: reg is a04c0004
Resetting the modules...
During Resetting the modules...
After resetting the modules...
Port 1: Installed -- AUTO FXS/DPO
Port 2: Installed -- AUTO FXO (UK mode)
Port 3: Not installed
Port 4: Not installed
VPM100: Not Present
Found a Wildcard TDM: Wildcard TDM410P (4 modules)
My zapata.conf looks like this:
[trunkgroups]
[channels]
usecallerid=yes
cidsignalling=v23
cidstart=polarity
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
;echotraining=yes
;echocancelwhenbridged=yes
immediate=no
faxdetect=no
fwringdetect=1
context=incoming
group=1
usecallerid=yes
faxdetect=none
signalling=fxs_ks
rxgain=8
txgain=8
callerid=asreceived
channel = 2
I've added a debug entry into my extension.conf:
[incoming]
exten = s,1,Verbose(Callerid = ${CALLERID} - ${CALLERIDNUM})
When I make an incoming call (I've got two landlines), I see this on my
terminal:
Asterisk Ready.
== Starting post polarity CID detection on channel 2
-- Starting simple switch on 'Zap/2-1'
[Apr 8 13:29:11] NOTICE[6815]: chan_zap.c:6171 ss_thread: Got event 2
(Ring/Answered)...
-- Executing [EMAIL PROTECTED]:1] Verbose(Zap/2-1, Callerid = - )
in new stack
Callerid = -
== Auto fallthrough, channel 'Zap/2-1' status is 'UNKNOWN'
-- Hungup 'Zap/2-1'
[Apr 8 13:29:13] NOTICE[6815]: cdr.c:434 ast_cdr_free: CDR on channel
'Zap/2-1' not posted
-- Starting simple switch on 'Zap/2-1'
-- Executing [EMAIL PROTECTED]:1] Verbose(Zap/2-1, Callerid = - )
in new stack
Callerid = -
== Auto fallthrough, channel 'Zap/2-1' status is 'UNKNOWN'
-- Hungup 'Zap/2-1'
[Apr 8 13:29:16] NOTICE[6816]: cdr.c:434 ast_cdr_free: CDR on channel
'Zap/2-1' not posted
== Starting post polarity CID detection on channel 2
-- Starting simple switch on 'Zap/2-1'
[Apr 8 13:29:28] WARNING[6817]: chan_zap.c:6234 ss_thread: CID timed
out waiting for ring. Exiting simple switch
-- Hungup 'Zap/2-1'
[Apr 8 13:29:28] NOTICE[6817]: cdr.c:434 ast_cdr_free: CDR on channel
'Zap/2-1' not posted
Executing last minute cleanups
The penultimate line appears after I hang up.
James
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Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
Tzafrir Cohen wrote: On Mon, Apr 07, 2008 at 06:11:02AM +0100, James Williamson wrote: Snap, Well, after trying to buying a TDM400P and then getting persuaded to buy a TDM410P because they no longer sell the 400 model I'd say I'm not impressed. It took three 2.6 kernel builds (zaptel 1.4 won't even build with the latest kernel release) What version have you tried? Of Zaptel and of the kernel? AFAIK 1.4.9.2 builds with latest kernel (or maybe there's actually a small warning with our drivers, fixed in SVN) Please provide an error log. Yes, zaptel 1.4.9.2 does compile against a 2.6.24.4 source tree, although the latest release on the website (something like a week ago) was 1.4.8 which doesn't: [EMAIL PROTECTED] zaptel-1.4.8]# make make[1]: Entering directory `/usr/local/src/zaptel-1.4.8' make -C /lib/modules/2.6.24/build SUBDIRS=/usr/local/src/zaptel-1.4.8 HOTPLUG_FIRMWARE=yes modules make[2]: Entering directory `/usr/src/linux-2.6.24' WARNING: Symbol version dump /usr/src/linux-2.6.24/Module.symvers is missing; modules will have no dependencies and modversions. scripts/Makefile.build:46: *** CFLAGS was changed in /usr/local/src/zaptel-1.4.8/Makefile. Fix it to use EXTRA_CFLAGS. Stop. make[2]: *** [_module_/usr/local/src/zaptel-1.4.8] Error 2 make[2]: Leaving directory `/usr/src/linux-2.6.24' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/local/src/zaptel-1.4.8' make: *** [all] Error 2 James -- James Williamson www.nameonthe.net tel: (+44) 870 1657215 direct: (+44) 1491 413609 fax: (+44) 1491 413606 email: [EMAIL PROTECTED] 'Hosting Java since 1999' email disclaimer: http://www.nameonthe.net/disclaimer.jsp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UK POTS - Is there a better card than TDM400P available ?
Snap, Well, after trying to buying a TDM400P and then getting persuaded to buy a TDM410P because they no longer sell the 400 model I'd say I'm not impressed. It took three 2.6 kernel builds (zaptel 1.4 won't even build with the latest kernel release) before I finally got the kernel to build and recognise the device. Can't get callerid to work, have followed the instructions to the letter, does it really have to be this hard? Must say I'm tempted to just send the card back and forget about it and just use Cisco CallManager. Digium's support to be fair have been responsive but unable to help. James Hi, I am feeling very frustrated with the Digium TDM400P, I have 3 x FXS 1x FXO modules and I have tried various things and different versions of Asterisk and Zaptel to no avail. Clearly there are issues with this card, so I am wondering - is there a card out there that does the following without the inherent problems of the TDM400 ? I.e a card that can reliably do: UK CID Distinctive Ring Detection. Any pointers would be great. Thanks. ( or is there an Asterisk version and Zaptel version in the 1.4 branch that fixes these issues ? ) Matt Brown ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM410 Callerid UK
Hi all, Has anyone got any experience with getting a TDM410 to work with callerid in the UK, I've spent some time fiddling with the options but haven't made any headway. I've also contacted digium support who haven't been able to help either. Many thanks, James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call
Let me start by saying I have no cisco phones, and no idea how to configure them. I will speak to the use of asterisk behind a NAT'ing firewall, which I believe to be your setup. Asterisk is pretty picky about how SIP and RTP packets are handled by a NAT firewall. Basically you need to maintain the same udp port for incoming and outgoing udp packets. I will attempt to illustrate this from what I have seen with my OpenBSD firewall: The SIP UA sends a packet registration packet to the Asterisk server, giving a port number with which it will accept reply packets. Our NAT however is in the middle and does this. SIP UA Register my port: 5060 (or 2842) sent to Asterisk port 5060 (or 2842) Firewall, ah outgoing UDP connection, to Asterisk port 5060, I will choose a random high port and rewrite the packet with my IP and the UDP port SomeHighPort. Asterisk gets this and responds to the ORIGINAL port (5060) and the firewall expects this to arrive on the UDP port SomeHighPort. For some resion even if the port 5060 is forwarded to the SIP UA, this packet gets lost. So you need to tell your firewall to somehow use the same port as the UA for the re-written packets. On openbsd I use this directive: nat on $ext_if inet proto udp from any to any - ($ext_if) static-port (someone please comment on how to do this under linux for our other readers) which says do not rewrite UDP port numbers. This is also necessary for RTP to work properly. Some people have success with using the qualify= directive in sip.conf to keep the session alive in the firewall by sending packets before it times out. But I believe this to be a far better solution, as the path for the UDP packets always exists, alive or dead. And this works nice if you have the following setup, 1 Asterisk - NAT - (n) SIP UA Since SIP will setup our calls, and Asterisk will assign different RTP UDP ports for different calls to different SIP UA. -- Best regards, Scottmailto:[EMAIL PROTECTED] fwd: 253984 Friday, March 12, 2004, 11:34:51 AM, you wrote: A Ok.. Let me start by saying that SJPhone works fine through NAT and the A Cisco phones inside the internal network work fine also... It's just the A Cisco phones on the outside using NAT. A For Testing I opened the Firewall open on the IP for the * Server. I A have done, everything you recommended below, but still no go... When the A phone registers with port 2842? Not the standard 5060? Any ideas? I A believe this is where my problem sits... A Thanks, A -gcc A -Original Message- A From: [EMAIL PROTECTED] A [mailto:[EMAIL PROTECTED] On Behalf Of James A Sizemore A Posted At: Friday, March 12, 2004 9:03 AM A Posted To: Asterisk User Group A Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum A retries exceeded on call A Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum A retries exceeded on call A Make sure your using qualify=500 in the sip.conf along with nat=yes, A make sure any firewalls allow 5060 udp and tcp and random ports above A 1 in form your PBX. A If you have all that it should work. A AstGrp wrote: Yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Sizemore Posted At: Thursday, March 11, 2004 10:47 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: Re: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call You do have : nat_enable: 1 nat_received_processing: 1 On the Ciscos? AstGrp wrote: I am having a similar problem... I get the same message, but inbound calls can go through This is only Cisco phones that are behind A NAT. I have tried your recommendations from below, but still no luck.. User can make outbound calls, just can't receive any. Any ideas would be greatly appreciated.. I even tried to change the timeout value in chan_sip, but it just waits longer to fail.. Just dosen't seem to want to communicate... Thanks, gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Bittner Posted At: Tuesday, March 02, 2004 11:46 PM Posted To: A Asterisk User Group Conversation: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Subject: RE: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call Are you using Cisco phones. ? I had this issue with my cisco phones. I didn't had any issues with dropped calls. All I did to fix this was set a prefered_codex and set proxy_register to 0. I hope this helps. John Bittner Simlab.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of dkwok Sent: Wednesday, March 03, 2004 7:04 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] chan_sip.c:495 retrans_pkt: Maximum retries exceeded on call *CLI
Re: [Asterisk-Users] voicemail not working with mysql !!!
Hello atif, send an e-mail to [EMAIL PROTECTED] I know nothing about voicemail and mysql configuration -- Best regards, Scottmailto:[EMAIL PROTECTED] ___ 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] NAT, Asterisk and SIP service provider (sipgate.de)
Hello Oliver,
okay, this was not easy and will make a long e-mail that I will also CC
to the list. I will answer in English because it is my native language.
I lived in Germany for 2.5 years and can speak German okay,
however I will spare you all of the declination failures that I make
on a regular basis.
I have an OpenBSD NAT'ting firewall allowing asterisk to talk to
sipgate.de with outgoing calls working nicely, incoming is untested
but should work.
sipgate.de is provides their services using SIP, and asterisk can be a
SIP client, you probably know this. SIP service providers setup their
systems to support normal SIP clients and you need to make you
asterisk and firewall (the and firewall bit is perhaps the most
important) appear to be a normal SIP client at the UDP port level.
SIP uses UDP port 5060 as its call setup/control port and some UDP
ports for its RTP media stream. The RTP media stream ports are set in
the asterisk control file rtp.conf.
I analyzed the traffic at the port level using xten's x-lite SIP
client talking to sipgate and discovered that the firewall setup is
very important. If you use NAT, standard procedure is to take outgoing
connections and translate them using some random high port as the source
port. so:
SIP Client NAT Firewall - sipgate.de
int ip : UDP 5060 NAT to: ext ip UDP 645035sipgate.de UDP 5060
The NAT firewall then keeps this config and expects to route info back
from sipgate to the internal SIP client on UDP port 645035. However
sipgate and the RFC think that SIP clients should accept info on UDP
port 5060 so it sends info back to (ext ip) UDP port 5060 and the
firewall may route this but it is not part of the same connection and
so it seems to get lost somehow.
What needs to be done is to tell the firewall to route all connections
on UDP 5060 out using UDP port 5060. in OpenBSD the pf.conf extries
look like this:
/etc/pf.conf:
# outgoing UDP port 5060 connections use source port 5060 on firewall
nat on $ext_if inet proto udp from any port = 5060 to any - ($ext_if) port 5060
# incomming UDP port 5060 connections should go to my asterisk server
rdr pass on $ext_if proto udp from any to ($ext_if) port 5060 - $voip_box
#RTP MEDIA STREAM redirect.
rdr pass on $ext_if proto udp from any to any port :20001 - $voip_box port
:20001
When this works, and keep in mind that this is for OpenBSD (I am not
sure if linux can do this), then asterisk setup is as follows:
/etc/asterisk/sip.conf
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
externip = xxx.sjwilliamson.ca
localnet = 192.168.1.0 ; Internal NETWORK address
localmask = 255.255.255.0 ; Internal netmask
nat = yes
register = 8007163:[EMAIL PROTECTED]/8007163
[sipgate]
secret=xxx
username=8007163
fromuser=8007163
fromdomain=sipgate.net
type=friend
host=sipgate.de
nat=yes
;qualify=yes
dtmfmode=rfc2833
canreinvite=no
context=in-sipgate
/etc/asterisk/rtp.conf - this is stock
;
; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
rtpstart=1
rtpend=2
/etc/asterisk/extensions.conf
;outgoing sipgate calls
[sipgatede]
exten = _0049.,1,SetCallerID(4921158007163)
exten = _0049.,2,SetCIDName(Scott Williamson)
exten = _0049.,3,Dial(SIP/0${EXTEN:[EMAIL PROTECTED],30)
exten = _0049.,4,Playback(the-party-you-are-calling)
exten = _0049.,5,Playback(is-curntly-unavail)
exten = _0049.,6,Hangup
exten = _0049.,104,Playback(the-party-you-are-calling)
exten = _0049.,105,Playback(is-curntly-busy)
exten = _0049.,106,Wait,3
exten = _0049.,107,Hangup
;incomming sipgate calls
[in-sipgate]
exten = 8007163,1,Macro(stdexten,1234,${PHONE1})
Incomming calls in the context [in-sipgate] need to have an extension
that is the same as your sipgate number. And you need to register with
this also.
Good luck, and remember that in this case the firewall config is the
most important, second is the extension / sipgate number in the
registration and in the context [in-sipgate].
Also, show sip registry at the asterisk console will show if you
have registered with sipgate. They seem to go offline sometimes, and I
do not know why. I consider this to be normal, as this happens to
other SIP accounts that I have.
Scott Williamson
P.S. Maybe you can try calling me over sipgate @ +49 211 58 00 71 63 to test and
see if incoming calls work.
--
Best regards,
Scottmailto:[EMAIL PROTECTED]
---
|Toronto | +1 416 xxx | PSTN |
|-|---|-|
| Düsseldorf | +49 211 58 00 71 63 | International |
| London | +44 20 71 27 63 82 | PSTN ENUM|
|-|---|-|
|FWD | 25 39 84 | VOIP |
| iaxTel | 1 700 839 8593 | |
[Asterisk-Users] SIP Behind NAT (sipgate.de)
Hello Users, I am attempting to create a sip connection in the following network: Sipgate.de -- Internet -- Gate -- Asterisk PBX -- Some Extension Gate, the gateway and nat'ing firewall has sip udp (5060) traffic and rtm udp (8000 to 8020) traffic forwarded to the asterisk pbx machine. Both asterisk and x-lite are set to listen/use these port ranges. (The forwards work, as X-lite works perfectly when forced to no firewall (Open IP) setting, simply writing the correct sip via: headers. I tested it this morning by calling to a normal number in .de) However, when I attempt to use asterisk to do the same thing, I get some strange behaviour. I have attempted to use the following different configurations of sip.conf: I am using asterisk cvs version v1-0_stable and chan_sip.c from cvs is verson 1.292.2.6. config #1 (the nat config) --snip-- [general] port = 5060 bindaddr = 0.0.0.0 externip = gate.sjwilliamson.ca localnet = 192.168.1.0 localmask = 255.255.255.0 context = local nat=yes register = xxx:[EMAIL PROTECTED]/6464 ; 6464 is my internal extension [sipgate] type=friend secret=xxx username=xxx host=sipgate.de nat=yes --end snip-- config #2 (the no-nat config) --snip-- [general] port = 5060 bindaddr = 0.0.0.0 ;externip = gate.sjwilliamson.ca ;localnet = 192.168.1.0 ;localmask = 255.255.255.0 context = local ;nat=yes register = xxx:[EMAIL PROTECTED]/6464 ; 6464 is my internal extension [sipgate] type=friend secret=xxx username=xxx host=sipgate.de ;nat=yes --end snip-- What I find strage is: 1. When using the no-nat config (2) asterisk is able to register with sipgate.de even though it sends out my internal address in the sip via header. However any sip invites are wrongly tagged with my internal ip address, and sipgate.de does not send me any audio, and the call times out (as it should). See following snippit: --snip-- 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK22fc46b2 From: sip:[EMAIL PROTECTED];tag=as11899051 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 (no NAT) to 217.10.79.9:5060 Sip read: 0 headers, 0 lines Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK22fc46b2;rport=62600;received=24.102.192.227 From: sip:[EMAIL PROTECTED];tag=as11899051 To: sip:[EMAIL PROTECTED];tag=b11cb9bb270104b49a99a995b8c68544.0784 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER WWW-Authenticate: Digest realm=sipgate.de, nonce=x Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 217.10.79.9:5060 Noisy feedback tells: pid=14272 req_src_ip=24.102.192.227 req_src_port=62600 in_uri=sip:sipgate.de out_uri=sip:sipgate.de via_cnt==1 10 headers, 0 lines 12 headers, 0 lines Reliably Transmitting: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK22fc46b2 From: sip:[EMAIL PROTECTED];tag=as11899051 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Authorization: Digest username=8007163, realm=sipgate.de, algorithm=MD5, uri=sip:sipgate.de, nonce=x, response=x Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 (no NAT) to 217.10.79.9:5060 Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK22fc46b2;rport=62600;received=24.102.192.227 From: sip:[EMAIL PROTECTED];tag=as11899051 To: sip:[EMAIL PROTECTED];tag=b11cb9bb270104b49a99a995b8c68544.0784 Call-ID: [EMAIL PROTECTED] CSeq: 103 REGISTER Contact: sip:[EMAIL PROTECTED]:5060;q=0.00;expires=780 Contact: sip:[EMAIL PROTECTED]:62600;q=0.00;expires=120 Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 217.10.79.9:5060 Noisy feedback tells: pid=14265 req_src_ip=24.102.192.227 req_src_port=62600 in_uri=sip:sipgate.de out_uri=sip:sipgate.de via_cnt==1 11 headers, 0 lines --end snip-- 2. When I use the nat config (1) it gets even stranger. Asterisk cannot register with sipgate.de, even though the sip via header reflects my correct internet ip address. It attempts to re-transmit five times, with the interesting line (no NAT) to 217.10.79.9:5060. --snip-- 11 headers, 0 lines Reliably Transmitting: REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 24.102.192.227:5060;branch=z9hG4bK1cd66717 From: sip:[EMAIL PROTECTED];tag=as7c8d34b5 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER User-Agent: Asterisk PBX Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-length: 0 (no NAT) to 217.10.79.9:5060 Retransmitting #1 (no NAT): REGISTER sip:sipgate.de SIP/2.0 Via: SIP/2.0/UDP 24.102.192.227:5060;branch=z9hG4bK2b457839 From: sip:[EMAIL PROTECTED];tag=as06e7a197 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 104 REGISTER
Re: [Asterisk-Users] SIP Behind NAT (sipgate.de)
Hello List, Just thought I would post an update, I got asterisk to register with sipgate.de. I was wrong, it was my firewall (maybe). Here is the way a normal nat under openbsd pf works: udp 192.168.1.100:5060 - 24.102.192.227:(random port) - 217.10.79.9:5060 but add this line to pf.conf before your main catch all nat line: nat on $ext_if inet proto udp from any port = 5060 to any - ($ext_if) port 5060 this changes the nat behaviour to use choose a static port on the firewall to originate the connection from. (reading man pages is good) and you get this: udp 192.168.1.100:5060 - 24.102.192.227:5060 - 217.10.79.9:5060 Which makes sense, as they (sipgate.de) want to see, and reply to port 5060 on the asterisk machine. I guess x-lite is a bit of a smarter UA when it comes to nat connections. Anyway hope this helps someone! Scott Thursday, February 19, 2004, 2:23:00 PM, you wrote: SJW Hello Users, SJW I am attempting to create a sip connection in the following network: Sipgate.de -- Internet -- Gate -- Asterisk PBX -- Some Extension SJW Gate, the gateway and nat'ing firewall has sip udp (5060) traffic and SJW rtm udp (8000 to 8020) traffic forwarded to the asterisk pbx machine. SJW Both asterisk and x-lite are set to listen/use these port ranges. SJW (The forwards work, as X-lite works perfectly when forced to no SJW firewall (Open IP) setting, simply writing the correct sip via: headers. I SJW tested it this morning by calling to a normal number in .de) SJW However, when I attempt to use asterisk to do the same thing, I get SJW some strange behaviour. I have attempted to use the following SJW different configurations of sip.conf: SJW I am using asterisk cvs version v1-0_stable and SJW chan_sip.c from cvs is verson 1.292.2.6. SJW config #1 (the nat config) SJW --snip-- SJW [general] SJW port = 5060 SJW bindaddr = 0.0.0.0 SJW externip = gate.sjwilliamson.ca SJW localnet = 192.168.1.0 SJW localmask = 255.255.255.0 SJW context = local SJW nat=yes register = xxx:[EMAIL PROTECTED]/6464 ; 6464 is my internal extension SJW [sipgate] SJW type=friend SJW secret=xxx SJW username=xxx SJW host=sipgate.de SJW nat=yes SJW --end snip-- SJW config #2 (the no-nat config) SJW --snip-- SJW [general] SJW port = 5060 SJW bindaddr = 0.0.0.0 SJW ;externip = gate.sjwilliamson.ca SJW ;localnet = 192.168.1.0 SJW ;localmask = 255.255.255.0 SJW context = local SJW ;nat=yes register = xxx:[EMAIL PROTECTED]/6464 ; 6464 is my internal extension SJW [sipgate] SJW type=friend SJW secret=xxx SJW username=xxx SJW host=sipgate.de SJW ;nat=yes SJW --end snip-- SJW What I find strage is: SJW 1. When using the no-nat config (2) asterisk is able to register SJW with sipgate.de even though it sends out my internal address in SJW the sip via header. However any sip invites are wrongly tagged SJW with my internal ip address, and sipgate.de does not send me any SJW audio, and the call times out (as it should). See following SJW snippit: SJW --snip-- SJW 11 headers, 0 lines SJW Reliably Transmitting: SJW REGISTER sip:sipgate.de SIP/2.0 SJW Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK22fc46b2 SJW From: sip:[EMAIL PROTECTED];tag=as11899051 SJW To: sip:[EMAIL PROTECTED] SJW Call-ID: [EMAIL PROTECTED] SJW CSeq: 102 REGISTER SJW User-Agent: Asterisk PBX SJW Expires: 120 SJW Contact: sip:[EMAIL PROTECTED] SJW Event: registration SJW Content-length: 0 SJW (no NAT) to 217.10.79.9:5060 SJW Sip read: SJW 0 headers, 0 lines SJW Sip read: SJW SIP/2.0 401 Unauthorized SJW Via: SIP/2.0/UDP SJW 192.168.1.100:5060;branch=z9hG4bK22fc46b2;rport=62600;received=24.102.192.227 SJW From: sip:[EMAIL PROTECTED];tag=as11899051 SJW To: SJW sip:[EMAIL PROTECTED];tag=b11cb9bb270104b49a99a995b8c68544.0784 SJW Call-ID: [EMAIL PROTECTED] SJW CSeq: 102 REGISTER SJW WWW-Authenticate: Digest realm=sipgate.de, nonce=x SJW Server: Sip EXpress router (0.8.12 (i386/linux)) SJW Content-Length: 0 SJW Warning: 392 217.10.79.9:5060 Noisy feedback tells: SJW pid=14272 req_src_ip=24.102.192.227 req_src_port=62600 SJW in_uri=sip:sipgate.de out_uri=sip:sipgate.de via_cnt==1 SJW 10 headers, 0 lines SJW 12 headers, 0 lines SJW Reliably Transmitting: SJW REGISTER sip:sipgate.de SIP/2.0 SJW Via: SIP/2.0/UDP 192.168.1.100:5060;branch=z9hG4bK22fc46b2 SJW From: sip:[EMAIL PROTECTED];tag=as11899051 SJW To: sip:[EMAIL PROTECTED] SJW Call-ID: [EMAIL PROTECTED] SJW CSeq: 103 REGISTER SJW User-Agent: Asterisk PBX SJW Authorization: Digest username=8007163, realm=sipgate.de, SJW algorithm=MD5, uri=sip:sipgate.de, nonce=x, response=x SJW Expires: 120 SJW Contact: sip:[EMAIL PROTECTED] SJW Event: registration SJW Content-length: 0 SJW (no NAT) to 217.10.79.9:5060 SJW Sip read: SJW SIP/2.0 200 OK SJW Via: SIP/2.0/UDP SJW 192.168.1.100:5060;branch=z9hG4bK22fc46b2;rport=62600;received=24.102.192.227 SJW From: sip:[EMAIL PROTECTED];tag=as11899051 SJW To: SJW sip:[EMAIL
