Re: [asterisk-users] SIP registration fails
SIP-registration errors are solved by restarting the Asterisk-server.
But I expect them to return in time...
I can make call now, but the other end does not hear me. So problem with
RTP-flow...
Can someone guide me to the solution ?
On the Asterisk-server I have this (iptables):
-A RH-Firewall-1-INPUT -p udp --dport 4569 -j ACCEPT
-A RH-Firewall-1-INPUT -p tcp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp --dport 5060 -j ACCEPT
-A RH-Firewall-1-INPUT -p udp --dport 11000:11500 -j ACCEPT
-A RH-Firewall-1-INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT
-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 25 -j
ACCEPT
-A RH-Firewall-1-INPUT -m state --state NEW -m tcp -p tcp --dport 22 -j
ACCEPT
-A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited
In rtp.conf I have this :
rtpstart=11000
rtpend=11500
Asterisk is behind firewall. Endian firewall has following
configuration :
enable SIP proxy transparant
RTP port low : 11000
RTP port high : 11500
Firewall port forwarding : uplink:5060 >>> asterisk_ip:5060
Asterisk himself says :
-- Executing [050510...@intern:1] NoOp("SIP/grandstream-09813b58",
"via 3StarsNet") in new stack
-- Executing [050510...@intern:2] Dial("SIP/grandstream-09813b58",
"SIP/3starsnet/050510484") in new stack
-- Called 3starsnet/050510484
-- SIP/3starsnet-0981bf08 is making progress passing it to
SIP/grandstream-09813b58
-- SIP/3starsnet-0981bf08 answered SIP/grandstream-09813b58
== Spawn extension (intern, 050510484, 2) exited non-zero on
'SIP/grandstream-09813b58'
What do I need in sip.conf to overcome these rtp-problems ??
I have :
externip=78.21.62.99
canreinvite=no
jbenable = yes
[3starsnet]
type=peer
...
nat=yes
...
Thanks for the help !
Jonas.
On Thu, 2009-06-25 at 17:25 +0200, jonas kellens wrote:
> Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports
> opened and 5060 forwarded to Asterisk (192.168.2.2)
>
> Can someone see why SIP-registration fails ??
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[asterisk-users] SIP registration fails
Asterisk-server behind Endian-firewall: SIP-aware, 5060 + RTP-ports opened and 5060 forwarded to Asterisk (192.168.2.2) Can someone see why SIP-registration fails ?? register => 092779077:x...@85.119.188.3 [3starsnet] type=peer host=85.119.188.3 username=092779077 secret= fromuser=092779077 fromdomain=sip.3starsnet.com dtmfmode=rfc2833 canreinvite=no insecure=port,invite qualify=yes nat=yes disallow=all allow=gsm allow=alaw [Jun 25 16:54:32] NOTICE[32550]: chan_sip.c:7683 sip_reg_timeout:-- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #54) Really destroying SIP dialog '628e05295c1a2cc560d1c6c073b85...@127.0.0.1' Method: REGISTER Really destroying SIP dialog '4f9b2b7a241f3f2a193ceb0020778...@192.168.2.2' Method: OPTIONS Retransmitting #4 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK15be023b;rport From: ;tag=as36b44350 To: Call-ID: 628e05295c1a2cc560d1c6c073b85...@127.0.0.1 CSeq: 156 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- Reliably Transmitting (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: "asterisk" ;tag=as6cd2d842 To: Contact: Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #1 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: "asterisk" ;tag=as6cd2d842 To: Contact: Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #5 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK15be023b;rport From: ;tag=as36b44350 To: Call-ID: 628e05295c1a2cc560d1c6c073b85...@127.0.0.1 CSeq: 156 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- Retransmitting #2 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: "asterisk" ;tag=as6cd2d842 To: Contact: Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #3 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: "asterisk" ;tag=as6cd2d842 To: Contact: Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Retransmitting #4 (NAT) to 85.119.188.3:5060: OPTIONS sip:sip.3starsnet.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK6243112d;rport From: "asterisk" ;tag=as6cd2d842 To: Contact: Call-ID: 0cf19c845a5a8a543c2b959718164...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 25 Jun 2009 14:54:42 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ___ -- 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] sip registration fails with 404
Can anyone give me any direction as to why I'm getting a 404 during the registration process. Sip Debug is: <-- SIP read from 192.168.99.110:5060: REGISTER sip:asterisk1.rightsolve.com SIP/2.0 Via: SIP/2.0/UDP 192.168.99.110:5060;branch=123456789 To: From: ;tag=12345 CSeq: 1 REGISTER Call-ID: f97f33fb6c6a82c2efc0a16258e93d6b Max-Forwards: 70 User-Agent: VCS Contact: Expires: 600 Content-Length: 0 --- (11 headers 0 lines)--- Using latest REGISTER request as basis request Sending to 192.168.99.110 : 5060 (non-NAT) Transmitting (no NAT) to 192.168.99.110:5060: SIP/2.0 404 Not found Via: SIP/2.0/UDP 192.168.99.110:5060;branch=123456789;received=192.168.99.110 From: ;tag=12345 To: ;tag=as5106c249 Call-ID: f97f33fb6c6a82c2efc0a16258e93d6b CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: Content-Length: 0 TIA, routerguy This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --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] SIP registration fails with realtime
I have set up realtime for Asterisk just as the instruction provide. Everything works, except it apearer that SIP devices do not regisert correctly. I can place a call from a SIP device, but not place a call to a SIP device. If a I use sip.conf everything seems to work. I have not posted all the configurations here because I'm just looking for a set of checks to follow. I noted that several other people on different lists have the same issue, but I have found no answer I understand.__Do You Yahoo!?Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
AW: [Asterisk-Users] SIP registration fails
Title: AW: [Asterisk-Users] SIP registration fails Hi Seshu, that's where I started off. But most of them are not working (at least not for me). My desired setup (for now) is very simple: SIP provider(web.de) <--> * <--> 2 SIP phones But none of the examples explains how the "register" statement and the corresponding host-entry are linked to each other. Could you help? Will -Ursprüngliche Nachricht- Von: Kanuri, Seshu (Company IT) [mailto:[EMAIL PROTECTED]] Gesendet: Mittwoch, 13. April 2005 20:11 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: RE: [Asterisk-Users] SIP registration fails You may better look at example sip.conf files you will be able to find on WIKI as there appears to be several incosnsistencies in your sip.conf. My suggestion is get rid off what you dont need and use only those what is barely essential. When you are using NAT Ip you need to have entries like: host=dynamic Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of William Marks Sent: Wednesday, April 13, 2005 10:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SIP registration fails Hello List ;) I'm quite amazed by the features, asterisk offers but as I'm quite new to this stuff, I've got a few questions. First of all the relevant part of my sip.conf: cut sip.conf -- [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to srvlookup=yes nat=yes localnet=192.168.11.0/255.255.255.0 externip= realm= context = from-sip ; Default for incoming calls insecure=very tos=0x18 dtmfmode=info disallow=all allow=gsm allow=alaw allow=ulaw register => :@sip.web.de/ [webde] type=friend username= secret= host=sip.web.de fromuser= fromdomain=sip.web.de nat=no canreinvite=no insecure=very qualify=400 dtmfmode=info cut sip.conf -- My questions on this are: a) why is SIP registration failing? b) how is mapping between "register=>" and [webde] done? many thanks. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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 registration fails
Title: SIP registration fails You may better look at example sip.conf files you will be able to find on WIKI as there appears to be several incosnsistencies in your sip.conf. My suggestion is get rid off what you dont need and use only those what is barely essential. When you are using NAT Ip you need to have entries like: host=dynamic Seshu From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of William MarksSent: Wednesday, April 13, 2005 10:57 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] SIP registration fails Hello List ;) I'm quite amazed by the features, asterisk offers but as I'm quite new to this stuff, I've got a few questions. First of all the relevant part of my sip.conf: cut sip.conf -- [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to srvlookup=yes nat=yes localnet=192.168.11.0/255.255.255.0 externip= realm= context = from-sip ; Default for incoming calls insecure=very tos=0x18 dtmfmode=info disallow=all allow=gsm allow=alaw allow=ulaw register => :@sip.web.de/ [webde] type=friend username= secret= host=sip.web.de fromuser= fromdomain=sip.web.de nat=no canreinvite=no insecure=very qualify=400 dtmfmode=info cut sip.conf -- My questions on this are: a) why is SIP registration failing? b) how is mapping between "register=>" and [webde] done? many thanks. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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] SIP registration fails
Title: SIP registration fails Hello List ;) I'm quite amazed by the features, asterisk offers but as I'm quite new to this stuff, I've got a few questions. First of all the relevant part of my sip.conf: cut sip.conf -- [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to srvlookup=yes nat=yes localnet=192.168.11.0/255.255.255.0 externip= realm= context = from-sip ; Default for incoming calls insecure=very tos=0x18 dtmfmode=info disallow=all allow=gsm allow=alaw allow=ulaw register => :@sip.web.de/ [webde] type=friend username= secret= host=sip.web.de fromuser= fromdomain=sip.web.de nat=no canreinvite=no insecure=very qualify=400 dtmfmode=info cut sip.conf -- My questions on this are: a) why is SIP registration failing? b) how is mapping between "register=>" and [webde] done? many thanks. ___ 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 registration fails
I have this problem for 2 days and i dont understand
I am behind a nat
my sip.conf is:
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-sip
disallow = all
allow= gsm
allow= ilbc
allow= ulaw
allow= alaw
;
;
localnet = 172.27.254.0/255.255.255.0 ; intern network ip address
;localmask = 255.255.255.0 ;
externip =193.49.116.12 ; my public ip address
;
maxexpirey=180
defaultexpirey=160
;
register => 560793:[EMAIL PROTECTED]/6002
;
[fwd]
type=friend
secret=mypasswd
username=fayafibun
host=fwd.pulver.com
fromdomain=fwd.pulver.com
insecure=very
context = from-sip
;
;
;
;
[bombaclaat]
callerid=("bombaclaat" <6009>)
type=friend
secret=mypasswd
host=dynamic
auth=md5
defaultip=172.27.254.14
context=internal
reinvite=no
canreinvite=no
dtmfmode=rfc2833
disallow=all
allow=all
mailbox=bombaclaat
qualify=1000
nat=yes
;
;
[6002]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
disallow=all
allow=all
;context=internal
context = from-sip
mailbox=6002
;
[6000]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
disallow=all
allow=all
context=internal
mailbox=6000
;
[bloodclaat]
type=friend
host=dynamic
reinvite=no
canreinvite=no
qualify=1000
dtmfmode=rfc2833
disallow=all
allow=all
context=internal
mailbox=bloodclaat
;
;
my extension.conf
[general]
static=yes
writeprotect=no
[globals]
;
; The name to use on callerid
;
BOMBA=SIP/bombaclaat
OTRE=SIP/6002
FWDUSERID=560793
FWDUSERNAME=fayafibun
PHONE1=6002
PHONE1VM=voicemail(6002)
FWDEXTEND=6002
;EVRYONE=${BOMBA}&${OTRE}
;
[internal]
;
; local extensions
;
exten => bombaclaat,1,Dial(SIP/bombaclaat,60) ; call SIP extension
"bombaclaat" for 60 seconds, if extension bombaclaat is called
exten => bombaclaat,2,Voicemail(ubombaclaat) ; if we cant connect
to "bombaclaat" or after seconds go to the unavail VM
exten => bombaclaat,102,Voicemail(bbombaclaat); if busy, go to the busy VM
exten => 6002,1,Dial(SIP/6002,60) ; call SIP extension
"bombaclaat" for 60 seconds, if extension bombaclaat is called
exten => 6002,2,Voicemail(u6002) ; if we cant connect
to "bombaclaat" or after seconds go to the unavail VM
exten => 6002,102,Voicemail(b6002); if busy, go to the busy VM
exten => bloodclaat,1,Dial(SIP/bloodclaat,60)
exten => bloodclaat,2,Voicemail(ubloodclaat)
exten => bloodclaat,103,Voicemail(bbloodclaat)
exten => 6000,1,Dial(SIP/6000,60)
exten => 6000,2,Voicemail(u6000)
exten => 6000,103,Voicemail(b6000)
exten => _[123456789],1,NoOp("callfor"${EXTEN})
exten => _[123456789],2,Dial(SIP/${EXTEN},40,tr)
exten => _[123456789],3,Congestion
exten => 1312605133,1,Dial(${FIPC}/${EXTEN:1},60) ; call SIP
extension "bombaclaat" for 60 seconds, if extensio$
exten => 1312605133,2,Voicemail(ubombaclaat) ; if we cant connect
to "bombaclaat" or after seconds go to t$
exten => bombaclaat,104,Voicemail(bbombaclaat);;
;
;appeler le 2500 de n importe kel phone pour contacter le voicemail system
exten => 2500,1,VoicemailMain
exten => 2500,2,Hangup
;
;
; Voicemail System
;
exten => 123,1,Answer
exten => 123,2,Playback(tt-weasels)
exten => 123,3,Voicemail(6002)
exten => 123,4,Hangup
;
;
;exten => ,1,VoiceMailMain(${CALLERIDNUM}) ; extension is
the VM system,
; go directly to callers VM
;exten => ,2,Hangup
;
;[outbound-internal]
;
; include local extensions
;
; include => internal
;
;
; include SIP accounts
;
; include => 6002
; include => bombaclaat
; include => 6000
; include => bloodclaat
[default]
;
; include from-sip for default. We dont use it, but it might be a good idea
;
;include => internal
;Extension Description
;101 Mark Spencer
;102 Wil Meadows
;0 Operator
include => from-sip
include => fwd-out
[fwd-out]
exten => _7.,1,SetCIDNum(${FWDUSERID})
exten => _7.,2,SetCIDName(${FWDUSERNAME})
exten => _7.,3,Dial(SIP/fwd-outgoin/${EXTEN:1})
exten => _7.,4,Playback(invalid)
exten => _7.,5,Hangup
[from-sip]
exten => ${FWDEXTEN},1,Dial(${PHONE1},30)
exten => ${FWDEXTEN},2,Voicemail(u${PHONE1VM})
exten => ${FWDEXTEN},3,Hangup
exten => ${FWDEXTEN},102,Voicemail(b${PHONE1VM})
exten => ${FWDEXTEN},103,Hangup
I have those errors
Jan 20 11:30:18 NOTICE[98310]: chan_sip.c:4053 sip_reg_timeout:
Registration for '
Re: [Asterisk-Users] sip registration fails
Alberto Martínez wrote: Hello, I am trying to register in asterisk with a softphone (x-lite) and I am getting the following message: Jan 19 13:27:01 NOTICE[3359]: chan_sip.c:7531 handle_request: Registration from 'tito ' failed for '192.168.1.5' Just a guess, but the ip's don't match up. [...] I get the following message too and I don't know what does that means: Jan 19 13:26:41 WARNING[3343]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) I'm getting this too. Using sip debug shows some sort of message notification attempt repeating itself for a sip client even though the client isn't online. The series of repeats ends with the error message that you are seeing. Dave CAUTION: This message and any attachments contain privileged and confidential information. If you are not the intended recipient of this message, you are hereby notified that any use, dissemination, distribution or reproduction of this message is prohibited. If you have received this message in error please notify the sender immediately via email and then destroy this message and any attachments. Any views expressed in this message are those of the individual sender and may not necessarily reflect the views of Winstone Pulp International Ltd. ___ 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 registration fails
I have tried uncommenting the section for xlite included in the sample
configuration file sip.conf and I can't register.
[xlite1]
;Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!
;Note that Xlite sends NAT keep-alive packets, so qualify=yes is not needed
type=friend
regexten=1234 ; When they register, create extension 1234
username=tito
callerid="yo" <5678>
host=dynamic
nat=yes ; X-Lite is behind a NAT router
canreinvite=no; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
AM> Hello,
AM> I am trying to register in asterisk with a softphone (x-lite) and I am
AM> getting the following message:
AM> Jan 19 13:27:01 NOTICE[3359]: chan_sip.c:7531 handle_request:
AM> Registration from 'tito ' failed for
AM> '192.168.1.5'
AM> In the sip.conf file I have included the following. Does I need to
AM> include another change to allow the user to register?
AM> [phone1]
AM> type=friend
AM> host=dynamic
AM> defaultip=192.168.1.5
AM> username=tito
AM> secret=tito
AM> dtmfmode=rfc2833
AM> mailbox=1000
AM> context=sip
AM> callerid="Tito" <2124>
AM> I get the following message too and I don't know what does that means:
AM> Jan 19 13:26:41 WARNING[3343]: chan_sip.c:685 retrans_pkt:
AM> Maximum retries exceeded on call
AM> [EMAIL PROTECTED] for seqno 102
AM> (Non-critical Request)
AM> ___
AM> Asterisk-Users mailing list
AM> Asterisk-Users@lists.digium.com
AM> http://lists.digium.com/mailman/listinfo/asterisk-users
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AM>http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] sip registration fails
Hello, I am trying to register in asterisk with a softphone (x-lite) and I am getting the following message: Jan 19 13:27:01 NOTICE[3359]: chan_sip.c:7531 handle_request: Registration from 'tito ' failed for '192.168.1.5' In the sip.conf file I have included the following. Does I need to include another change to allow the user to register? [phone1] type=friend host=dynamic defaultip=192.168.1.5 username=tito secret=tito dtmfmode=rfc2833 mailbox=1000 context=sip callerid="Tito" <2124> I get the following message too and I don't know what does that means: Jan 19 13:26:41 WARNING[3343]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) ___ 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] SIP registration fails
Thanks for the info so far. I am still trying to asterisk'ize my ML9.2 firewall box and can't get the external SIP registration to work. If I hook up my Sipura directly to the WAN it registers OK. This is the message I get from asterisk: Mar 8 21:03:07 NOTICE[196621]: chan_sip.c:3140 sip_reg_timeout: Registration for '[EMAIL PROTECTED]' timed out, trying again If tried all combinations of firewall and asterisk settings (as well as turning the firewall completely off). I don't understand why this would now work. ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 192.168.1.1 ; Address to bind to localnet = 192.168.1.0 ; Internal network address localmask = 255.255.255.0 ; Internal netmask outside_addr = a.b.c.d; External network, a.b.c.d is my external IP externip = a.b.c.d; Public IP ; register=263872:[EMAIL PROTECTED]/263872 ; [fwd] type=friend secret=mypass username=263872 host=fwd.pulver.com My shorewall/rules I used looks like this (pretty open, rtp.conf was adjusted accordingly): ACCEPT masqfw tcp domain,bootps,http,https,631,imap,pop3,smtp,nntp,50600:50610 ACCEPT masqfw udp domain,bootps,http,https,631,imap,pop3,smtp,nntp,50600:50610 ACCEPT fw masqtcp 631,515,137,138,139,50600:50610 - ACCEPT fw masqudp 631,515,137,138,139,50600:50610 - ACCEPT loc fw tcp 22,5060:5070,50600:50610 - ACCEPT loc fw udp 5060:5070,50600:50610 - ACCEPT net loc udp 5060:5070,50600:50610 - ACCEPT net fw tcp 22,4662,2234,6882,6346,,,5060:5070,50600:50610 ACCEPT net fw udp 4666,2234,6882,6346,,,5060:5070,50600:50610 Could it be a port conflict with the Sipura unit on the LAN? The two lines register OK from the LAN on the 5060 and 5061 ports: -- SIP Seeding '2201' at [EMAIL PROTECTED]:5060 for 3600 -- SIP Seeding '2202' at [EMAIL PROTECTED]:5061 for 3600 ___ 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
