Can't make calls through betamax accounts

parseint32 · February 5, 2013, 4:11pm

I registered a telbo.com account some month ago, recently i created another one, but when i try to call with the second one asterisk prints:

== Using SIP RTP CoS mark 5 -- Executing [*******@casa:1] Dial("SIP/2001-0000002e", "SIP/*******@telbo,180,TtKkXxr") in new stack == Using SIP RTP CoS mark 5 -- Called SIP/*******@telbo == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/2001-0000002e' status is 'CHANUNAVAIL'

With x-lite both accounts works fine.

Yesterday i signed up to freevoipdeal.com, and i had the same problem… then i realized that there is some problem with my asterisk server.

sip.conf:

[general]
	context=chiamate-ingresso
	directmedia=no
	nat=force_rport,comedia
	localnet=127.0.0.1/255.0.0.0
	bindaddr=******
	bindport=****
	externip=******
	realm=*******
	srvlookup=yes
	allowguest=no
	alwaysauthreject=yes
	;-------------------------			CODECS			-------------------------
	disallow=all
	allow=alaw
	allow=ulaw
	allow=gsm
	;-------------------------	REGISTRTAZIONE PROVIDER	-------------------------	
	register => account1:*****@sip.freevoipdeal.com:5060/account1
	register => anotheraccount:*****@sip.telbo.com/anotheraccount

[freevoipdeal]
	type=peer
	context=chiamate-uscita
	username=anotheraccount
	fromuser=anotheraccount
	secret=*****
	host=sip.freevoipdeal.com
	fromdomain=sip.freevoipdeal.com
	qualify=yes
	insecure=invite,port

[telbo]
	type=peer
	context=chiamate-uscita
	username=account1
	fromuser=account1
	secret=*****
	host=sip.telbo.com
	fromdomain=sip.telbo.com
	qualify=yes
	insecure=invite,port	

...
other accounts
...

The strange thing is that the old one account works!

ifconfig result:

lo Link encap:Local Loopback inet addr:127.0.0.1 Mask:255.0.0.0 inet6 addr: ::1/128 Scope:Host UP LOOPBACK RUNNING MTU:16436 Metric:1 RX packets:13100657 errors:0 dropped:0 overruns:0 frame:0 TX packets:13100657 errors:0 dropped:0 overruns:0 carrier:0 collisions:0 txqueuelen:0 RX bytes:135266732 (129.0 MiB) TX bytes:135266732 (129.0 MiB)

I do not use 5060 port

dejanst · February 6, 2013, 11:13am

Is your ifconfig command output copy/paste correct or did you make a mistake? You will not be able to access the internet with only a loopback interface …

We need more details for solving this problem. Enable SIP debugging in Asterisk (sip set debug on), make an outgoing call and copy/paste the command output.

parseint32 · February 6, 2013, 11:32am

Complete ifconfig:

[code]eth0 Link encap:Ethernet HWaddr 00:16:3C:E1:2E:5B
inet addr:serverpublicaddr Bcast:serverpublicaddr Mask:255.255.252.0
inet6 addr: fe80::216:3cff:fee1:2e5b/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:1174092698 errors:0 dropped:27479 overruns:0 frame:0
TX packets:2691891094 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:2671076793 (2.4 GiB) TX bytes:2333508039 (2.1 GiB)
Interrupt:10

eth1 Link encap:Ethernet HWaddr 00:18:3E:F1:B3:BC
inet addr:10.0.2.87 Bcast:10.0.15.255 Mask:255.255.240.0
inet6 addr: fe80::218:3eff:fef1:b3bc/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:33745269 errors:0 dropped:0 overruns:0 frame:0
TX packets:13 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:1601552871 (1.4 GiB) TX bytes:830 (830.0 b)
Interrupt:11 Base address:0x4000

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:19367325 errors:0 dropped:0 overruns:0 frame:0
TX packets:19367325 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:643281526 (613.4 MiB) TX bytes:643281526 (613.4 MiB)
[/code]

debug:

---
Reliably Transmitting (no NAT) to 77.72.174.128:5060:
OPTIONS sip:sip.freevoipdeal.com SIP/2.0
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK5333abfe
Max-Forwards: 70
From: "asterisk" <sip:3febbraio2013@SERVERPUBLICADDR:ASTERISKPORT>;tag=as2977308e
To: <sip:sip.freevoipdeal.com>
Contact: <sip:3febbraio2013@SERVERPUBLICADDR:ASTERISKPORT>
Call-ID: 4ed8696e2919effa1d1efae92b6991ec@SERVERPUBLICADDR:ASTERISKPORT
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.1.2
Date: Wed, 06 Feb 2013 11:28:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:83.211.227.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK5dfa3657
From: "asterisk" <sip:0917481522@SERVERPUBLICADDR:ASTERISKPORT>;tag=as7172476d
To: <sip:voip.eutelia.it>;tag=c040a69dfc7733bdec8c921a7a9f2d3a.352f
Call-ID: 48fced665332a90b28cf80990443589f@SERVERPUBLICADDR:ASTERISKPORT
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: SPS EUT RM GW 04
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '48fced665332a90b28cf80990443589f@SERVERPUBLICADDR:ASTERISKPORT' Method: OPTIONS

<--- SIP read from UDP:77.72.174.128:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK5333abfe
From: "asterisk" <sip:3febbraio2013@SERVERPUBLICADDR:ASTERISKPORT>;tag=as2977308e
To: <sip:sip.freevoipdeal.com>
Contact: sip:77.72.174.128:5060
Call-ID: 4ed8696e2919effa1d1efae92b6991ec@SERVERPUBLICADDR:ASTERISKPORT
CSeq: 102 OPTIONS
Supported: foo
User-Agent: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Accept: application/sdp

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '4ed8696e2919effa1d1efae92b6991ec@SERVERPUBLICADDR:ASTERISKPORT' Method: OPTIONS

<--- SIP read from UDP:HOMEADDR:1056 --->
ACK sip:CALLEDNUMBER@SERVERPUBLICADDR:ASTERISKPORT SIP/2.0
Via: SIP/2.0/UDP 192.168.0.50:5061;branch=z9hG4bK-59a551bf
From: "Filippo Casa" <sip:2001@SERVERPUBLICADDR>;tag=928e9975b1f48252o1
To: <sip:CALLEDNUMBER@SERVERPUBLICADDR>;tag=as3447e9aa
Call-ID: a0ce0671-61c59986@192.168.0.50
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="2001",realm="sip.epd-clan.it",nonce="6a8e3bff",uri="sip:CALLEDNUMBER@SERVERPUBLICADDR:ASTERISKPORT",algorithm=MD5,response="a121b879eeb82b895ad53b1d3e4d6a0c"
Contact: "Filippo Casa" <sip:2001@192.168.0.50:5061;ref=2001>
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog 'a0ce0671-61c59986@192.168.0.50' Method: ACK
Reliably Transmitting (no NAT) to 212.97.59.76:5061:
OPTIONS sip:sip.messagenet.it SIP/2.0
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK2ef79ffe
Max-Forwards: 70
From: "asterisk" <sip:5298507@SERVERPUBLICADDR:ASTERISKPORT>;tag=as663b5bb3
To: <sip:sip.messagenet.it>
Contact: <sip:5298507@SERVERPUBLICADDR:ASTERISKPORT>
Call-ID: 397eafb42c31781f60e89e3e7c18ade3@SERVERPUBLICADDR:ASTERISKPORT
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.1.2
Date: Wed, 06 Feb 2013 11:28:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:212.97.59.76:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK2ef79ffe
From: "asterisk" <sip:5298507@SERVERPUBLICADDR:ASTERISKPORT>;tag=as663b5bb3
To: <sip:sip.messagenet.it>;tag=98df93dff07c6bc0cd4e22344f9aa5a7.7386
Call-ID: 397eafb42c31781f60e89e3e7c18ade3@SERVERPUBLICADDR:ASTERISKPORT
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding:
Accept-Language: en
Supported:
Server: sip.messagenet.it SIP Proxy
Content-Length: 0
Warning: 392 212.97.59.76:5061 "Noisy feedback tells: pid=26871 req_src_ip=SERVERPUBLICADDR req_src_port=ASTERISKPORT in_uri=sip:sip.messagenet.it out_uri=sip:sip.messagenet.it via_cnt==1"

<------------->
--- (13 headers 0 lines) ---
Really destroying SIP dialog '397eafb42c31781f60e89e3e7c18ade3@SERVERPUBLICADDR:ASTERISKPORT' Method: OPTIONS
Reliably Transmitting (no NAT) to 77.72.174.128:5060:
OPTIONS sip:sip.telbo.com SIP/2.0
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK6e520cba
Max-Forwards: 70
From: "asterisk" <sip:cartmatina@SERVERPUBLICADDR:ASTERISKPORT>;tag=as039e4657
To: <sip:sip.telbo.com>
Contact: <sip:cartmatina@SERVERPUBLICADDR:ASTERISKPORT>
Call-ID: 1177fa10143461f653c8bdc26f955e5e@SERVERPUBLICADDR:ASTERISKPORT
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.1.2
Date: Wed, 06 Feb 2013 11:28:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
Reliably Transmitting (NAT) to HOMEADDR:48482:
OPTIONS sip:2100@192.168.0.41:50994;transport=udp SIP/2.0
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK4af0089b;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@SERVERPUBLICADDR:ASTERISKPORT>;tag=as294e3745
To: <sip:2100@192.168.0.41:50994;transport=udp>
Contact: <sip:asterisk@SERVERPUBLICADDR:ASTERISKPORT>
Call-ID: 55f58e07610934f2012487923fbdc72f@SERVERPUBLICADDR:ASTERISKPORT
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.1.2
Date: Wed, 06 Feb 2013 11:28:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:77.72.174.128:5060 --->
SIP/2.0 200 Ok
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK6e520cba
From: "asterisk" <sip:cartmatina@SERVERPUBLICADDR:ASTERISKPORT>;tag=as039e4657
To: <sip:sip.telbo.com>
Contact: sip:77.72.174.128:5060
Call-ID: 1177fa10143461f653c8bdc26f955e5e@SERVERPUBLICADDR:ASTERISKPORT
CSeq: 102 OPTIONS
Supported: foo
User-Agent: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
Accept: application/sdp

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '1177fa10143461f653c8bdc26f955e5e@SERVERPUBLICADDR:ASTERISKPORT' Method: OPTIONS

<--- SIP read from UDP:HOMEADDR:48482 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK4af0089b;rport=ASTERISKPORT
To: <sip:2100@192.168.0.41:50994;transport=udp>
From: "asterisk" <sip:asterisk@SERVERPUBLICADDR:ASTERISKPORT>;tag=as294e3745
Call-ID: 55f58e07610934f2012487923fbdc72f@SERVERPUBLICADDR:ASTERISKPORT
CSeq: 102 OPTIONS
Contact: <sip:2100@192.168.0.41:50994;transport=udp>
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '55f58e07610934f2012487923fbdc72f@SERVERPUBLICADDR:ASTERISKPORT' Method: OPTIONS

<--- SIP read from UDP:HOMEADDR:1055 --->
NOTIFY sip:SERVERPUBLICADDR:ASTERISKPORT SIP/2.0
Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK-cd74e929
From: "Casa" <sip:2000@SERVERPUBLICADDR>;tag=fa46686a28bedfebo0
To: <sip:SERVERPUBLICADDR>
Call-ID: af3136be-317045a7@192.168.0.50
CSeq: 47721 NOTIFY
Max-Forwards: 70
Contact: "Casa" <sip:2000@192.168.0.50:5060;ref=2000>
Event: keep-alive
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- Transmitting (no NAT) to HOMEADDR:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK-cd74e929;received=HOMEADDR
From: "Casa" <sip:2000@SERVERPUBLICADDR>;tag=fa46686a28bedfebo0
To: <sip:SERVERPUBLICADDR>;tag=as5161e38a
Call-ID: af3136be-317045a7@192.168.0.50
CSeq: 47721 NOTIFY
Server: Asterisk PBX 11.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'af3136be-317045a7@192.168.0.50' in 32000 ms (Method: NOTIFY)

<--- SIP read from UDP:HOMEADDR:1056 --->
NOTIFY sip:SERVERPUBLICADDR:ASTERISKPORT SIP/2.0
Via: SIP/2.0/UDP 192.168.0.50:5061;branch=z9hG4bK-ccd1539a
From: "Filippo Casa" <sip:2001@SERVERPUBLICADDR>;tag=fa4a4c72e1490b83o1
To: <sip:SERVERPUBLICADDR>
Call-ID: 35b9dbd2-fedfc3a3@192.168.0.50
CSeq: 47721 NOTIFY
Max-Forwards: 70
Contact: "Filippo Casa" <sip:2001@192.168.0.50:5061;ref=2001>
Event: keep-alive
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---

<--- Transmitting (no NAT) to HOMEADDR:5061 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.50:5061;branch=z9hG4bK-ccd1539a;received=HOMEADDR
From: "Filippo Casa" <sip:2001@SERVERPUBLICADDR>;tag=fa4a4c72e1490b83o1
To: <sip:SERVERPUBLICADDR>;tag=as0ce4a266
Call-ID: 35b9dbd2-fedfc3a3@192.168.0.50
CSeq: 47721 NOTIFY
Server: Asterisk PBX 11.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '35b9dbd2-fedfc3a3@192.168.0.50' in 32000 ms (Method: NOTIFY)
game-state*CLI> sip set debug off
SIP Debugging Disabled

77.72.174.128 is sip.freevoipdeal.com ip.

i can’t copy the previous messages because cli prints the lines too fast!

dejanst · February 6, 2013, 11:35am

We need a call trace. That means something that starts with SIP Invite message

parseint32 · February 6, 2013, 11:38am

i did it:

[code]<------------>
– Executing [CALLEDNUMBER@casa:1] Dial(“SIP/2001-00000060”, “SIP/0039CALLEDNUMBER@freevoipdeal,180,TtKkXxr”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 15868
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 77.72.174.128:5060:
INVITE sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060 SIP/2.0
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK21beef2a
Max-Forwards: 70
From: “Filippo Casa” sip:3febbraio2013@sip.freevoipdeal.com;tag=as3b3f7853
To: sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060
Contact: sip:3febbraio2013@SERVERPUBLICADDR:ASTERISKPORT
Call-ID: 31b3ec4026213bfd61e166744abb36da@sip.freevoipdeal.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.1.2
Date: Wed, 06 Feb 2013 11:36:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 896793141 896793141 IN IP4 SERVERPUBLICADDR
s=Asterisk PBX 11.1.2
c=IN IP4 SERVERPUBLICADDR
t=0 0
m=audio 15868 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

-- Called SIP/0039CALLEDNUMBER@freevoipdeal

<— Transmitting (NAT) to HOMEADDR:1056 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.50:5061;branch=z9hG4bK-e58bdd15;received=HOMEADDR;rport=1056
From: “Filippo Casa” sip:2001@SERVERPUBLICADDR;tag=98ce69543b191feao1
To: sip:CALLEDNUMBER@SERVERPUBLICADDR;tag=as0bbc79d6
Call-ID: ccfced1c-91932ec2@192.168.0.50
CSeq: 102 INVITE
Server: Asterisk PBX 11.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:CALLEDNUMBER@SERVERPUBLICADDR:ASTERISKPORT
Content-Length: 0

<------------>

<— SIP read from UDP:77.72.174.128:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK21beef2a
From: “Filippo Casa” sip:3febbraio2013@sip.freevoipdeal.com;tag=as3b3f7853
To: sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060
Contact: sip:0039CALLEDNUMBER@77.72.174.128:5060
Call-ID: 31b3ec4026213bfd61e166744abb36da@sip.freevoipdeal.com
CSeq: 102 INVITE
Server: (Very nice Sip Registrar/Proxy Server)
Allow: ACK,BYE,CANCEL,INVITE,REGISTER,OPTIONS,INFO,MESSAGE
WWW-Authenticate: Digest realm=“sip.freevoipdeal.com”,nonce=“2414804031”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Transmitting (no NAT) to 77.72.174.128:5060:
ACK sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060 SIP/2.0
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK21beef2a
Max-Forwards: 70
From: “Filippo Casa” sip:3febbraio2013@sip.freevoipdeal.com;tag=as3b3f7853
To: sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060
Contact: sip:3febbraio2013@SERVERPUBLICADDR:ASTERISKPORT
Call-ID: 31b3ec4026213bfd61e166744abb36da@sip.freevoipdeal.com
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.1.2
Content-Length: 0

Audio is at 15868
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 77.72.174.128:5060:
INVITE sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060 SIP/2.0
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK0352ff35
Max-Forwards: 70
From: “Filippo Casa” sip:3febbraio2013@sip.freevoipdeal.com;tag=as3b3f7853
To: sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060
Contact: sip:3febbraio2013@SERVERPUBLICADDR:ASTERISKPORT
Call-ID: 31b3ec4026213bfd61e166744abb36da@sip.freevoipdeal.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX 11.1.2
Authorization: Digest username=“3febbraio2013”, realm=“sip.freevoipdeal.com”, algorithm=MD5, uri=“sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060”, nonce=“2414804031”, response="c2cb2b61ea15c42062f51350bf9a2c42"
Date: Wed, 06 Feb 2013 11:36:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282

v=0
o=root 896793141 896793142 IN IP4 SERVERPUBLICADDR
s=Asterisk PBX 11.1.2
c=IN IP4 SERVERPUBLICADDR
t=0 0
m=audio 15868 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<— SIP read from UDP:77.72.174.128:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK0352ff35
From: “Filippo Casa” sip:3febbraio2013@sip.freevoipdeal.com;tag=as3b3f7853
To: sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060
Contact: sip:0039CALLEDNUMBER@77.72.174.128:5060
Call-ID: 31b3ec4026213bfd61e166744abb36da@sip.freevoipdeal.com
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) —

<— SIP read from UDP:77.72.174.128:5060 —>
SIP/2.0 404 User not found
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK0352ff35
From: “Filippo Casa” sip:3febbraio2013@sip.freevoipdeal.com;tag=as3b3f7853
To: sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060
Contact: sip:0039CALLEDNUMBER@77.72.174.128:5060
Call-ID: 31b3ec4026213bfd61e166744abb36da@sip.freevoipdeal.com
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) —
Transmitting (no NAT) to 77.72.174.128:5060:
ACK sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060 SIP/2.0
Via: SIP/2.0/UDP SERVERPUBLICADDR:ASTERISKPORT;branch=z9hG4bK0352ff35
Max-Forwards: 70
From: “Filippo Casa” sip:3febbraio2013@sip.freevoipdeal.com;tag=as3b3f7853
To: sip:0039CALLEDNUMBER@sip.freevoipdeal.com:5060
Contact: sip:3febbraio2013@SERVERPUBLICADDR:ASTERISKPORT
Call-ID: 31b3ec4026213bfd61e166744abb36da@sip.freevoipdeal.com
CSeq: 103 ACK
User-Agent: Asterisk PBX 11.1.2
Content-Length: 0

Scheduling destruction of SIP dialog ‘31b3ec4026213bfd61e166744abb36da@sip.freevoipdeal.com’ in 6400 ms (Method: INVITE)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/2001-00000060’ status is ‘CHANUNAVAIL’

<— Reliably Transmitting (NAT) to HOMEADDR:1056 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.50:5061;branch=z9hG4bK-e58bdd15;received=HOMEADDR;rport=1056
From: “Filippo Casa” sip:2001@SERVERPUBLICADDR;tag=98ce69543b191feao1
To: sip:CALLEDNUMBER@SERVERPUBLICADDR;tag=as0bbc79d6
Call-ID: ccfced1c-91932ec2@192.168.0.50
CSeq: 102 INVITE
Server: Asterisk PBX 11.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0

<------------>

<— SIP read from UDP:HOMEADDR:1056 —>
ACK sip:CALLEDNUMBER@SERVERPUBLICADDR:ASTERISKPORT SIP/2.0
Via: SIP/2.0/UDP 192.168.0.50:5061;branch=z9hG4bK-e58bdd15
From: “Filippo Casa” sip:2001@SERVERPUBLICADDR;tag=98ce69543b191feao1
To: sip:CALLEDNUMBER@SERVERPUBLICADDR;tag=as0bbc79d6
Call-ID: ccfced1c-91932ec2@192.168.0.50
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username=“2001”,realm=“sip.epd-clan.it”,nonce=“1ccaf723”,uri=“sip:CALLEDNUMBER@SERVERPUBLICADDR:ASTERISKPORT”,algorithm=MD5,response="df24bd52daad36b9d758b298c690c59f"
Contact: “Filippo Casa” sip:2001@192.168.0.50:5061;ref=2001
User-Agent: Cisco/SPA122-1.2.1(004)
Allow-Events: talk, hold, conference
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Really destroying SIP dialog ‘ccfced1c-91932ec2@192.168.0.50’ Method: ACK
Really destroying SIP dialog ‘6165f2b65d765c325506fb7f6a9ecc01@sip.freevoipdeal.com’ Method: INVITE

<— SIP read from UDP:HOMEADDR:48482 —>

[/code]

parseint32 · February 6, 2013, 5:38pm

I added allow=all to [freevoipdeal].
Now when i call:

== Using SIP RTP CoS mark 5 -- Executing [callednumber@casa:1] Dial("SIP/2001-00000002", "SIP/callednumber@freevoipdeal,180,TtKkXxr") in new stack == Using SIP RTP CoS mark 5 failed to extend from 1024 to 1299 failed to extend from 1024 to 1311 failed to extend from 1024 to 1303 -- Called SIP/callednumber@freevoipdeal [Feb 6 12:36:50] WARNING[12269]: chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on transmission e0887b8-421de22c@192.168.0.50 for seqno 101 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6399ms with no response [Feb 6 12:36:53] WARNING[12269]: chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on transmission 4f8561f96037e3af1bc2ac9d073165d3@sip.freevoipdeal.com for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions Packet timed out after 6400ms with no response [Feb 6 12:36:53] WARNING[12269]: chan_sip.c:4193 retrans_pkt: Hanging up call 4f8561f96037e3af1bc2ac9d073165d3@sip.freevoipdeal.com - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions). == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/2001-00000002' status is 'CHANUNAVAIL'

Debug says: SIP/2.0 503 Service Unavailable asterisk

ambiorixg12 · February 6, 2013, 6:21pm

this could help on your sip.conf

Change the User-Agent option from Asterisk PBX 11.1.2 to X-Lite

set insecure=port,invite

david55 · February 6, 2013, 6:40pm

insecure=invite only relates to incoming calls. insecure=port,invite is usually wrong, as it is more insecure than necessary. As I understand it, with modern versions of Asterisk, most cases should be unnecessary, as remotesecret is a cleaner way of dealing with the situation that insecure=invite works round.

If the ITSP is generating appropriate SIP responses, the problem is that it does not consider 0039CALLEDNUMBER to be a valid telephone number. Maybe it is expecting +39CALLEDNUMBER.

parseint32 · February 6, 2013, 7:08pm

it doesn’t work yet…

ambiorixg12 · February 6, 2013, 7:13pm

[quote=“david55”]insecure=invite only relates to incoming calls. insecure=port,invite is usually wrong, as it is more insecure than necessary. As I understand it, with modern versions of Asterisk, most cases should be unnecessary, as remotesecret is a cleaner way of dealing with the situation that insecure=invite works round.

If the ITSP is generating appropriate SIP responses, the problem is that it does not consider 0039CALLEDNUMBER to be a valid telephone number. Maybe it is expecting +39CALLEDNUMBER.[/quote]

just changed to insecure=invite and it finnaly worked!

parseint32 · February 6, 2013, 7:14pm

[quote=“ambiorixg12”][quote=“david55”]insecure=invite only relates to incoming calls. insecure=port,invite is usually wrong, as it is more insecure than necessary. As I understand it, with modern versions of Asterisk, most cases should be unnecessary, as remotesecret is a cleaner way of dealing with the situation that insecure=invite works round.

If the ITSP is generating appropriate SIP responses, the problem is that it does not consider 0039CALLEDNUMBER to be a valid telephone number. Maybe it is expecting +39CALLEDNUMBER.[/quote]

just changed to insecure=invite and it finnaly worked![/quote]
i edited my post… i was wrong, i was dialing with another provider

ambiorixg12 · February 6, 2013, 7:17pm

It Works or not ? and if works what was exactly the problem

parseint32 · February 6, 2013, 7:30pm

it actually doesn’t work: when i try to dial from one of bellmont site i can hear the busy tone.
If i try through xlite it works.
I tried to change useragent to “X-Lite”, and to set + instead 00, but didn’t worked

sip.conf


[general]
	context=chiamate-ingresso

	directmedia=no
	nat=force_rport,comedia
	localnet=127.0.0.1/255.0.0.0
	localnet=10.0.15.255/255.255.240.0
	bindaddr=serverpublicaddr
	bindport=asteriskport
	externip=serverpublicaddr
	realm=sip.epd-clan.it
	srvlookup=yes

	allowguest=no
	alwaysauthreject=yes

	disallow=all
	allow=alaw
	allow=ulaw
	allow=gsm
	register => **:**@sip.freevoipdeal.com:/**

..

[freevoipdeal]
	type=peer
	context=chiamate-uscita
	username=**
	fromuser=**
	secret=**
	host=sip.freevoipdeal.com
	fromdomain=sip.freevoipdeal.com
	qualify=yes
	insecure=invite ; if i remove it, nothing changes
	port=5060
	allow=all[/code]

On extensions.conf under my home context:
[code]exten => MYMOBILENUMBER,1,Dial(SIP/+39${EXTEN}@freevoipdeal,180,TtKkXxr)[/code]

When i dial:
[code]  == Using SIP RTP CoS mark 5
    -- Executing [MYMOBILENUMBER@casa:1] Dial("SIP/2001-00000006", "SIP/+39MYMOBILENUMBER@freevoipdeal,180,TtKkXxr") in new stack
  == Using SIP RTP CoS mark 5
failed to extend from 1024 to 1299
failed to extend from 1024 to 1311
failed to extend from 1024 to 1303
    -- Called SIP/+39MYMOBILENUMBER@freevoipdeal
[Feb  6 14:22:48] WARNING[17121]: chan_sip.c:4164 retrans_pkt: Retransmission timeout reached on transmission 0513439b618c6fc648e09f657c1a34b3@sip.freevoipdeal.com for seqno 102 (Critical Request) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[Feb  6 14:22:48] WARNING[17121]: chan_sip.c:4193 retrans_pkt: Hanging up call 0513439b618c6fc648e09f657c1a34b3@sip.freevoipdeal.com - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/2001-00000006' status is 'CHANUNAVAIL'

debug:

[code]<------------>
Really destroying SIP dialog ‘05fa993b313197fd71448585262dde46@sip.freevoipdeal.com’ Method: INVITE
Retransmitting #1 (NAT) to 31.216.135.115:1056:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.50:5061;branch=z9hG4bK-6bb94f36;received=31.216.135.115;rport=1056
From: “Filippo Casa” sip:2001@serverpublicaddr;tag=491896944c1e06c9o1
To: sip:MYMOBILENUMBER@serverpublicaddr;tag=as58187faf
Call-ID: fff6ceb0-c2d9cd5d@192.168.0.50
CSeq: 102 INVITE
Server: Asterisk PBX 11.1.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: No user responding
X-Asterisk-HangupCauseCode: 18
Content-Length: 0

[/code]

EDIT:

now it works… i do not understand what was wrong…
Since i set ulaw first it started to work.
Now i set ALL as the first post and it works

dejanst · February 7, 2013, 12:28pm

I would say that this definitly is something with the provider. Because the provider rejects the call, I don’t see anything wrong with the call setup. That is before the provider rejects the call .

My advice would be - take it up with the VoIP provider.

parseint32 · June 27, 2013, 12:04am

Today i tried another freevoipdeal account, and i had the same problem.

It doesn’t make sense… i tried 2 different account with their softphone and all worked fine, also with xlite they worked.
If i use asterisk, the new account doesn’t work. I also tried to use Xlite useragent, but with the same result.

How is it possible this happen? Is asterisk keeping a sort of connection alive?

dejanst · June 27, 2013, 10:22am

What if you ask the freevoipdeal people if they have a sample of Asterisk config?

If you want us to help you, for start please post a part of sip.conf wehere your freevoipdeal account is defined.

david55 · June 27, 2013, 10:28am

That’s what register does!

parseint32 · June 27, 2013, 1:11pm

That’s what register does![/quote]
So also xlite does it…
…how do they detect if users are registering from asterisk?

freevoipdeal.com/feedback no email

my new sip.conf

[code]freevoipdeal
type=peer
context=chiamate-uscita
host=sip.freevoipdeal.com
fromdomain=sip.freevoipdeal.com
qualify=yes
insecure=invite

account1
username=usr1
fromuser=usr1
secret=**

account2
username=usr2
fromuser=usr2
secret=** [/code]

anyway it is a betamax clone… where can i find an official configuration for asterisk 11?

dejanst · June 28, 2013, 1:15pm

You are missing the “register =>” statement in sip.conf. Google arround a bit and you should find the appropriate config where Asterisk is doing outbound registration.

parseint32 · June 28, 2013, 11:27pm

I forgot to paste them here…

Both account are registered. sip show registry:

Host                                    dnsmgr Username       Refresh State                Reg.Time

sip.freevoipdeal.com:5060               N      usr1       105 Registered           Sat, 29 Jun 2013 01:26:17
sip.freevoipdeal.com:5060               N      usr2       105 Registered           Sat, 29 Jun 2013 01:26:17

[...]

I contacted freevoipdeal staff… i’m waiting for their response