SIP response 500 Internal Server error

Asterisk Asterisk Support
1

For some time I’ve been using betamax Voipraider.
Since a few days, I can’t connect anymore.
During a call setup I get:

– Executing [002348068338xxx@default:14] Dial(“Zap/1-1”, “SIP/voipraiderqueen-out/002348068338xxx”) in new stack
– Called voipraiderqueen-out/002348068338xxx
– Got SIP response 500 “Internal server error” back from 194.221.62.198 <---------- :unamused:
– SIP/voipraiderqueen-out-08213720 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [002348068338xxx@default:15] Congestion(“Zap/1-1”, “”) in new stack
== Spawn extension (default, 002348068338xxx, 15) exited non-zero on ‘Zap/1-1’
– Hungup ‘Zap/1-1’

Also when using a softphone (SJphone) I get error message:

Server Internal error
Call rejected:500 Internal server error. <------ :unamused:

I contacted Voipraider helpdesk and got following response:
[ul]
This is a hardware problem, most likely with your modem or router. Please contact your supplier on this matter.

Kind regards,
Customer service
[/ul]

For me it’s difficult to believe that it is a hardware problem, because when I use other betamax providers such as voipbuster and Poivy, I do NOT get these error messages. Neither with Asterisk nor with a softphone.

Only Voipraider is (now) causing this problem.
(b.t.w. voipraider works fine with their own client.)

How can help me out? :laughing:

2

Is the “SIP 500 Internal server error” a fault with me or with Voipraider? :unamused:

3

Enable sip debugging and post your logs.

4

sip.conf.

[ul]
;register to voipraider
register => hoegema1946:mysecret@sip.Voipraider.com

[voipraider-out]
type=peer
username=hoegema1946
secret=mysecret
host=sip.Voipraider.com
realm=sip.Voipraider.com
fromuser=00324763788xx
context=default
canreinvite=no
insecure=very
qualify=300
nat=yes
port=5060
dtmfmode=inband
disallow=all
allow=alaw
allow=ulaw
[/ul]

routes-outgoing.conf

[ul]
exten => _000234.,1,NoOP(Time=${STRFTIME(${EPOCH},%H)}.${STRFTIME(${EPOCH},%M)})
exten => _000234.,n,Dial(SIP/voipraider-out/${EXTEN:1})
exten => _000234.,n,Congestion()
[/ul]

This is CLI output with debug off.:
– Starting simple switch on ‘Zap/1-1’
– Executing [0002348023042515@default:1] NoOp(“Zap/1-1”, “Time=16.53”) in new stack
– Executing [0002348023042515@default:2] Dial(“Zap/1-1”, “SIP/voipraider-out/002348023042515”) in new stack
– Called voipraider-out/002348023042515
– Got SIP response 500 “Internal server error” back from 194.221.62.198
– SIP/voipraider-out-0820fef0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [0002348023042515@default:3] Congestion(“Zap/1-1”, “”) in new stack
== Spawn extension (default, 0002348023042515, 3) exited non-zero on ‘Zap/1-1’
– Hungup ‘Zap/1-1’

asterisk*CLI> sip set debug peer voipraider-out
SIP Debugging Enabled for IP: 194.221.62.198:5060
– Saved useragent “Linksys/SPA3000-3.1.18(GW)” for peer 1000
– Starting simple switch on ‘Zap/1-1’
Really destroying SIP dialog ‘50b7cc0a3a501fd636f08b88613b7bea@127.0.0.1’ Method: REGISTER
Really destroying SIP dialog ‘576571311a8020826f299e33539143ac@127.0.0.1’ Method: REGISTER
– Executing [0002348023042515@default:1] NoOp(“Zap/1-1”, “Time=16.28”) in new stack
– Executing [0002348023042515@default:2] Dial(“Zap/1-1”, “SIP/voipraider-out/002348023042515”) in new stack
Audio is at 78.20.154.229 port 5094
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (NAT) to 194.221.62.198:5060:
INVITE sip:002348023042515@sip.Voipraider.com SIP/2.0
Via: SIP/2.0/UDP 78.20.154.229:5060;branch=z9hG4bK715f739e;rport
From: “1205” sip:0032476378861@78.20.154.229;tag=as4cded629
To: sip:002348023042515@sip.Voipraider.com
Contact: sip:0032476378861@78.20.154.229
Call-ID: 6f0794c81422e304131e0bbe68648317@78.20.154.229
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 18 Dec 2007 16:28:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 2545 2545 IN IP4 78.20.154.229
s=session
c=IN IP4 78.20.154.229
t=0 0
m=audio 5094 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

-- Called voipraider-out/002348023042515

asterisk*CLI>
<— SIP read from 194.221.62.198:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 78.20.154.229:5060;branch=z9hG4bK715f739e;rport
From: “1205” sip:0032476378861@78.20.154.229;tag=as4cded629
To: sip:002348023042515@sip.Voipraider.com
Contact: sip:002348023042515@194.221.62.198:5060
Call-ID: 6f0794c81422e304131e0bbe68648317@78.20.154.229
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm=“sipdiscount.com”,nonce=“2892617360”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Transmitting (NAT) to 194.221.62.198:5060:
ACK sip:002348023042515@sip.Voipraider.com SIP/2.0
Via: SIP/2.0/UDP 78.20.154.229:5060;branch=z9hG4bK715f739e;rport
From: “1205” sip:0032476378861@78.20.154.229;tag=as4cded629
To: sip:002348023042515@sip.Voipraider.com
Contact: sip:0032476378861@78.20.154.229
Call-ID: 6f0794c81422e304131e0bbe68648317@78.20.154.229
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Audio is at 78.20.154.229 port 5094
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (NAT) to 194.221.62.198:5060:
INVITE sip:002348023042515@sip.Voipraider.com SIP/2.0
Via: SIP/2.0/UDP 78.20.154.229:5060;branch=z9hG4bK202b472e;rport
From: “1205” sip:0032476378861@78.20.154.229;tag=as4cded629
To: sip:002348023042515@sip.Voipraider.com
Contact: sip:0032476378861@78.20.154.229
Call-ID: 6f0794c81422e304131e0bbe68648317@78.20.154.229
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“hoegema1946”, realm=“sipdiscount.com”, algorithm=MD5, uri="sip:002348023042515@sip.Voipraider.com", nonce=“2892617360”, response=“e6947274efffe34eb539ff87209e8c01”, opaque=“”
Date: Tue, 18 Dec 2007 16:28:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 207

v=0
o=root 2545 2546 IN IP4 78.20.154.229
s=session
c=IN IP4 78.20.154.229
t=0 0
m=audio 5094 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

asterisk*CLI>
<— SIP read from 194.221.62.198:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 78.20.154.229:5060;branch=z9hG4bK202b472e;rport
From: “1205” sip:0032476378861@78.20.154.229;tag=as4cded629
To: sip:002348023042515@sip.Voipraider.com
Contact: sip:002348023042515@194.221.62.198:5060
Call-ID: 6f0794c81422e304131e0bbe68648317@78.20.154.229
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
asterisk*CLI>
<— SIP read from 194.221.62.198:5060 —>
SIP/2.0 500 Internal server error
Via: SIP/2.0/UDP 78.20.154.229:5060;branch=z9hG4bK202b472e;rport
From: “1205” sip:0032476378861@78.20.154.229;tag=as4cded629
To: sip:002348023042515@sip.Voipraider.com
Contact: sip:002348023042515@194.221.62.198:5060
Call-ID: 6f0794c81422e304131e0bbe68648317@78.20.154.229
CSeq: 103 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Content-Length: 0

------------->
— (10 headers 0 lines) —
– Got SIP response 500 “Internal server error” back from 194.221.62.198
Transmitting (NAT) to 194.221.62.198:5060:
ACK sip:002348023042515@sip.Voipraider.com SIP/2.0
Via: SIP/2.0/UDP 78.20.154.229:5060;branch=z9hG4bK202b472e;rport
From: “1205” sip:0032476378861@78.20.154.229;tag=as4cded629
To: sip:002348023042515@sip.Voipraider.com
Contact: sip:0032476378861@78.20.154.229
Call-ID: 6f0794c81422e304131e0bbe68648317@78.20.154.229
CSeq: 103 ACK
ser-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

-- SIP/voipraider-out-0820fef0 is circuit-busy

== Everyone is busy/congested at this time (1:0/1/0)
– Executing [0002348023042515@default:3] Congestion(“Zap/1-1”, “”) in new stack
Really destroying SIP dialog ‘6f0794c81422e304131e0bbe68648317@78.20.154.229’ Method: INVITE
== Spawn extension (default, 0002348023042515, 3) exited non-zero on ‘Zap/1-1’
– Hungup ‘Zap/1-1’
asterisk*CLI>

I don’t have the right knowledge to do an interpretation of this output. :confused:

Who can help:

5

Noboddy who can help? :unamused:

I’ve not been able to find a solution myself. :cry:

6

I have been experiencing the same issue recently. Looking out for solutions.

7

With which provider?

I have tested at another site, with a new account name (using softphone SJPhone) and the result is the same: Internal server error.

Also tested with a hardphone at another site and the same result.

So I am (nearly) convinced the the error is not coming from my site.

Still looking for solution(s) :unamused:

8

Same: voipraider.com.
Also noticed that the same error occurs if I try to ‘register’ with this server (using register=> xxx@sip.voipraider.com).
Did not have this problem with voipwise.com (another betamax company). Neither with any other provider I use, including acanac, stanaphone, grandcentral etc). ‘lowratevoip.com’ should also work as I can register with them.

As a planB, I will consume my remaining credit of $11 by using their softphone, and then ditch it altogether.

9

[quote=“fractalspace”]Same: voipraider.com.
As a planB, I will consume my remaining credit of $11 by using their softphone, and then ditch it altogether.[/quote]

I was thinking of doing the same, but as a challenge I still want to investigate further. :laughing:

10

Just so you know, the advertized rates are applicable only when you use their client. Rates are different when using a SIP client.

11

I have been using Asterisk and not noticed any difference in prizing.

12

Can somebody try to register with Voipraider and configure a softphone (NOT voipraider’s own client) and see if you can make a call.

I have tried from different locations, and have NOT been able to call. Get always error message 500.

(it works ok with their own software client phone. but with asterisk or my own softphone (SJPhone) I can’t get it working)

13

Hi,
I tried with X-lite softphone and Im getting the same error “Internal Server Error” also I tried with asterisk and doesnt work. Im in México.

I have a public IP so I think this error isnt a nat problem.

I had a similar problem with another provider in México and I configured the asterisk with the next line (sip.conf):

[general]
useragent=name of the client that the provider gives me

This is because the provider only allows to connect to it with their software and with this the provider “thinks” that Im trying to connect with their softphone. However I dont know what agent I have to put in this case (voipraider)

Sorry for my english

14

[quote=“joako”]Hi,
I tried with X-lite softphone and Im getting the same error “Internal Server Error” also I tried with asterisk and doesnt work. Im in México.

I have a public IP so I think this error isnt a nat problem.

I had a similar problem with another provider in México and I configured the asterisk with the next line (sip.conf):

[general]
useragent=name of the client that the provider gives me

This is because the provider only allows to connect to it with their software and with this the provider “thinks” that Im trying to connect with their softphone. However I dont know what agent I have to put in this case (voipraider)

Sorry for my english[/quote]

Thanks for trying. :laughing:

I actually tried with:
[ul]
useragent=VoipRaider 4.01build 476
and
useragent="VoipRaider 4.01build 476"
and
useragent=VoipRaider4.01build476
[/ul]
Nothing of the above worked.

I did not put useragent under
[general]
but under
[voipraider-out]
type=peer
username=hoegema1946
secret=mysecret
host=sip.voipraider.com
realm=voipraider.com
fromdomain=sip.voipraider.com
fromuser=00324763788xx
useragent=VoipRaider4.01build476
context=default
canreinvite=no
insecure=invite
qualify=300
nat=yes
port=5060
dtmfmode=inband
disallow=all
allow=alaw
allow=ulaw

According to their website
voipraider.com/en/sipp.html
they should support SIP devices other than their own.

The funny thing is that it has been working well (some time ago now)

15

Since today Voipraider is working again. :laughing:

NOTHING was changed on my site, since it stopped working. :unamused: