I have registered in sip.conf and in my network i am not using any port
forwarding kind of stuff (NAT), Asterisk server is directly connected to
Internet and the Internet router doesn't have any firewall.
And attached is asterisk log, that SIP REGISTER messages keep on sending
and no response from the server.
I am sure that this is some network issue, because the same account i
tested in different network (Network B) in some other place and it got
registered, even i am able to make call.
One thing which i don't understand is in same network (Network A) in xlite
phone the account is getting registered and not in Asterisk server.
I just want to isolate things why I am not getting any response, or
somewhere the response is getting lost! :(
Regards,
Gopal.
On Wed, Sep 26, 2012 at 6:32 PM, SamyGo <govoi...@gmail.com> wrote:
> Hi,
> How are you connected to server ? How have you configured your asterisk
> server to register to other side ? What about any NAT involved in your
> scenario ?Turn on sip debug and share your registrations.
>
> BR
> Sammy
> On Sep 26, 2012 5:54 PM, "Danny Nicholas" <da...@debsinc.com> wrote:
>
>> Another possibility – you registered from the softphone first and the
>> provider took the IP address from your PC and “locked out” the IP address
>> of your Asterisk server.****
>>
>> ** **
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
>> *Sent:* Wednesday, September 26, 2012 7:51 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] SIP Retransmitting REGISTER message****
>>
>> ** **
>>
>> there is no firewall, its just the router gave by the service provider.
>> May be the SIP port issue?****
>>
>> ** **
>>
>> Regards.****
>>
>> ** **
>>
>> On Wed, Sep 26, 2012 at 6:17 PM, Danny Nicholas <da...@debsinc.com>
>> wrote:****
>>
>> The Asterisk server and softphone are hitting the firewall from two
>> different points. Start there.****
>>
>> ****
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
>> *Sent:* Wednesday, September 26, 2012 7:45 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] SIP Retransmitting REGISTER message****
>>
>> ****
>>
>> Hi,****
>>
>> ****
>>
>> I was trying to register a VoIP trunk in Asterisk , where its keep on
>> sending Register message to the server, where I am not getting any response
>> from server. ****
>>
>> ****
>>
>> But whereas if i register in Xlite softphone the account is getting
>> registered. ****
>>
>> ****
>>
>> I suspect it could be network related issue, but since in softphone it is
>> getting registered from the same network. ****
>>
>> ****
>>
>> Any ideas to isolate things would be appreciated. ****
>>
>> ****
>>
>> Regards,****
>>
>> Gopal. ****
>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> 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 --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>> http://www.asterisk.org/hello
>>
>> 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 --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
Nov 29 16:10:17 VERBOSE[5249] logger.c: Retransmitting #1 (NAT) to
202.85.243.105:5060:
REGISTER sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK566dab4d;rport
From: <sip:8889191...@sip.pennytel.com>;tag=as107f0d7d
To: <sip:8889191...@sip.pennytel.com>
Call-ID: 7163c7526c3e0a70262c658d7d527...@1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:Pennytel@117.223.64.95>
Event: registration
Content-Length: 0
---
Nov 29 16:10:17 VERBOSE[5249] logger.c: Retransmitting #1 (NAT) to
204.74.213.5:5061:
REGISTER sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK27a09fca;rport
From: <sip:905871...@sip.callwithus.com>;tag=as614f747c
To: <sip:905871...@sip.callwithus.com>
Call-ID: 63649a661edcb5861eb8188844452...@1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:905871851@117.223.64.95>
Event: registration
Content-Length: 0
---
Nov 29 16:10:17 VERBOSE[5249] logger.c: Retransmitting #2 (NAT) to
202.85.243.105:5060:
OPTIONS sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK03d0202f;rport
From: "asterisk" <sip:asterisk@117.223.64.95>;tag=as65e99acb
To: <sip:sip.pennytel.com>
Contact: <sip:asterisk@117.223.64.95>
Call-ID: 3eebd6d80f71c0431cae56e730a02...@1234.com
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 30 Nov 1999 04:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Nov 29 16:10:17 VERBOSE[5249] logger.c: Retransmitting #3 (NAT) to
202.85.243.105:5060:
REGISTER sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK21979700;rport
From: <sip:8889191...@sip.pennytel.com>;tag=as4b0bc6c9
To: <sip:8889191...@sip.pennytel.com>
Call-ID: 766cfc6c092d2a6165c79346578bf...@1234.com
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:Pennytel@117.223.64.95>
Event: registration
Content-Length: 0
---
Nov 29 16:10:17 VERBOSE[5249] logger.c: Retransmitting #3 (NAT) to
204.74.213.5:5061:
REGISTER sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK2c4ffad0;rport
From: <sip:905871...@sip.callwithus.com>;tag=as0c6cbedb
To: <sip:905871...@sip.callwithus.com>
Call-ID: 3d067e066f11c70448136f50014b9...@1234.com
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:905871851@117.223.64.95>
Event: registration
Content-Length: 0
---
Nov 29 16:10:18 VERBOSE[5249] logger.c: Retransmitting #2 (NAT) to
202.85.243.105:5060:
REGISTER sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK566dab4d;rport
From: <sip:8889191...@sip.pennytel.com>;tag=as107f0d7d
To: <sip:8889191...@sip.pennytel.com>
Call-ID: 7163c7526c3e0a70262c658d7d527...@1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:Pennytel@117.223.64.95>
Event: registration
Content-Length: 0
---
Nov 29 16:10:18 VERBOSE[5249] logger.c: Retransmitting #2 (NAT) to
204.74.213.5:5061:
REGISTER sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK27a09fca;rport
From: <sip:905871...@sip.callwithus.com>;tag=as614f747c
To: <sip:905871...@sip.callwithus.com>
Call-ID: 63649a661edcb5861eb8188844452...@1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:905871851@117.223.64.95>
Event: registration
Content-Length: 0
---
Nov 29 16:10:18 VERBOSE[5249] logger.c: Retransmitting #3 (NAT) to
202.85.243.105:5060:
OPTIONS sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK03d0202f;rport
From: "asterisk" <sip:asterisk@117.223.64.95>;tag=as65e99acb
To: <sip:sip.pennytel.com>
Contact: <sip:asterisk@117.223.64.95>
Call-ID: 3eebd6d80f71c0431cae56e730a02...@1234.com
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 30 Nov 1999 04:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Nov 29 16:10:19 VERBOSE[5249] logger.c: Retransmitting #4 (NAT) to
202.85.243.105:5060:
OPTIONS sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK03d0202f;rport
From: "asterisk" <sip:asterisk@117.223.64.95>;tag=as65e99acb
To: <sip:sip.pennytel.com>
Contact: <sip:asterisk@117.223.64.95>
Call-ID: 3eebd6d80f71c0431cae56e730a02...@1234.com
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 30 Nov 1999 04:10:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Nov 29 16:10:19 VERBOSE[5249] logger.c: Called :chan_sip.c:sip_destroy_nolock:
Nov 29 16:10:19 VERBOSE[5248] logger.c: Am HERE in pbx.c ast_hint_state_changed
Nov 29 16:10:20 VERBOSE[5249] logger.c: Retransmitting #3 (NAT) to
202.85.243.105:5060:
REGISTER sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK566dab4d;rport
From: <sip:8889191...@sip.pennytel.com>;tag=as107f0d7d
To: <sip:8889191...@sip.pennytel.com>
Call-ID: 7163c7526c3e0a70262c658d7d527...@1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:Pennytel@117.223.64.95>
Event: registration
Content-Length: 0
---
Nov 29 16:10:20 VERBOSE[5249] logger.c: Retransmitting #3 (NAT) to
204.74.213.5:5061:
REGISTER sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK27a09fca;rport
From: <sip:905871...@sip.callwithus.com>;tag=as614f747c
To: <sip:905871...@sip.callwithus.com>
Call-ID: 63649a661edcb5861eb8188844452...@1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:905871851@117.223.64.95>
Event: registration
Content-Length: 0
---
Nov 29 16:10:21 VERBOSE[5249] logger.c: Retransmitting #4 (NAT) to
202.85.243.105:5060:
REGISTER sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK21979700;rport
From: <sip:8889191...@sip.pennytel.com>;tag=as4b0bc6c9
To: <sip:8889191...@sip.pennytel.com>
Call-ID: 766cfc6c092d2a6165c79346578bf...@1234.com
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:Pennytel@117.223.64.95>
Event: registration
Content-Length: 0
---
Nov 29 16:10:21 VERBOSE[5249] logger.c: Retransmitting #4 (NAT) to
204.74.213.5:5061:
REGISTER sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK2c4ffad0;rport
From: <sip:905871...@sip.callwithus.com>;tag=as0c6cbedb
To: <sip:905871...@sip.callwithus.com>
Call-ID: 3d067e066f11c70448136f50014b9...@1234.com
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:905871851@117.223.64.95>
Event: registration
Content-Length: 0
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
