[asterisk-users] SIP Retransmitting REGISTER message

Thu Sep 27 02:09:32 CDT 2012

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 <govoiper at 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" <danny at 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-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at 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 <danny at debsinc.com>
>> wrote:****
>>
>> The Asterisk server and softphone are hitting the firewall from two
>> different points.  Start there.****
>>
>>  ****
>>
>> *From:* asterisk-users-bounces at lists.digium.com [mailto:
>> asterisk-users-bounces at 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
>
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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:8889191252 at sip.pennytel.com>;tag=as107f0d7d
To: <sip:8889191252 at sip.pennytel.com>
Call-ID: 7163c7526c3e0a70262c658d7d527f14 at 1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:Pennytel at 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:905871851 at sip.callwithus.com>;tag=as614f747c
To: <sip:905871851 at sip.callwithus.com>
Call-ID: 63649a661edcb5861eb8188844452195 at 1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:905871851 at 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 at 117.223.64.95>;tag=as65e99acb
To: <sip:sip.pennytel.com>
Contact: <sip:asterisk at 117.223.64.95>
Call-ID: 3eebd6d80f71c0431cae56e730a02658 at 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:8889191252 at sip.pennytel.com>;tag=as4b0bc6c9
To: <sip:8889191252 at sip.pennytel.com>
Call-ID: 766cfc6c092d2a6165c79346578bf364 at 1234.com
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:Pennytel at 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:905871851 at sip.callwithus.com>;tag=as0c6cbedb
To: <sip:905871851 at sip.callwithus.com>
Call-ID: 3d067e066f11c70448136f50014b9f9b at 1234.com
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:905871851 at 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:8889191252 at sip.pennytel.com>;tag=as107f0d7d
To: <sip:8889191252 at sip.pennytel.com>
Call-ID: 7163c7526c3e0a70262c658d7d527f14 at 1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:Pennytel at 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:905871851 at sip.callwithus.com>;tag=as614f747c
To: <sip:905871851 at sip.callwithus.com>
Call-ID: 63649a661edcb5861eb8188844452195 at 1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:905871851 at 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 at 117.223.64.95>;tag=as65e99acb
To: <sip:sip.pennytel.com>
Contact: <sip:asterisk at 117.223.64.95>
Call-ID: 3eebd6d80f71c0431cae56e730a02658 at 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 at 117.223.64.95>;tag=as65e99acb
To: <sip:sip.pennytel.com>
Contact: <sip:asterisk at 117.223.64.95>
Call-ID: 3eebd6d80f71c0431cae56e730a02658 at 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:8889191252 at sip.pennytel.com>;tag=as107f0d7d
To: <sip:8889191252 at sip.pennytel.com>
Call-ID: 7163c7526c3e0a70262c658d7d527f14 at 1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:Pennytel at 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:905871851 at sip.callwithus.com>;tag=as614f747c
To: <sip:905871851 at sip.callwithus.com>
Call-ID: 63649a661edcb5861eb8188844452195 at 1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:905871851 at 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:8889191252 at sip.pennytel.com>;tag=as4b0bc6c9
To: <sip:8889191252 at sip.pennytel.com>
Call-ID: 766cfc6c092d2a6165c79346578bf364 at 1234.com
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:Pennytel at 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:905871851 at sip.callwithus.com>;tag=as0c6cbedb
To: <sip:905871851 at sip.callwithus.com>
Call-ID: 3d067e066f11c70448136f50014b9f9b at 1234.com
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: <sip:905871851 at 117.223.64.95>
Event: registration
Content-Length: 0