SNOM 320 => ast_append_ha: NULL is not a valid IP

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prochmi

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hallo,
ich habe hier ein SNOM 320 und asterisk 1.2 (sip realtime)

ich kann das phone von anderen phones (cisco) aus anrufen.
wenn ich vom snom phone aus andere phones (oder auch die voicebox) anrufen will, bekomme ich am SNOM die meldung "not found: <nummer>" und im asterisk CLI "ast_append_ha: NULL is not a valid IP".

registrieren tut er sich problemlos.

irgendwelche ideen?

mfg,
michael
 
Kannst du mal bitte nen 'sip debug' machen und die Ausgabe der CLI inkl. debug hier posten?
 
danke, hat sich erledigt, ich habe bei den SIP einstellungen im SNOM phone fälschlicherweise eine landes und regionalvorwahl eingetragen. die hat er dann immer vorne angehängt, daher hat er die nummer nicht gefunden.

mfg,
michael
 
hallo,
leider muß ich das thema nochmal aufrollen.

durch die veränderten einstellungen kann ich zwar telefonieren, die fehlermeldung kommt aber trotzdem noch, hatte also nix miteinander zu tun.

hier der debug output:

Code: 
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2006.01.18 09:58:00 =~=~=~=~=~=~=~=~=~=~=~=


voip*CLI> 

voip*CLI> 

voip*CLI> 

voip*CLI> 

voip*CLI> 

voip*CLI> 

<-- SIP read from 192.168.1.7:5060: 
INVITE sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-q8684bcfh27w;rport

From: "VOIP2" <sip:[email protected]>;tag=truoz6lvpq

To: <sip:[email protected];user=phone>

Call-ID: 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:[email protected]:5060;line=ge0d7jnc>;flow-id=1

P-Key-Flags: keys="3"

User-Agent: snom320/5.0

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Content-Type: application/sdp

Content-Length: 366



v=0

o=root 1072681106 1072681106 IN IP4 192.168.1.7

s=call

c=IN IP4 192.168.1.7

t=0 0

m=audio 10740 RTP/AVP 0 8 9 2 3 18 4 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv



--- (17 headers 17 lines)---
Using INVITE request as basis request - 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9
Sending to 192.168.1.7 : 5060 (non-NAT)
Jan 18 08:51:55 WARNING[12241]: acl.c:182 ast_append_ha: NULL is not a valid IP
Jan 18 08:51:55 WARNING[12241]: acl.c:182 ast_append_ha: NULL is not a valid IP
Reliably Transmitting (NAT) to 192.168.1.7:5060:
SIP/2.0 407 Proxy Authentication Required

Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-q8684bcfh27w;received=192.168.1.7;rport=5060

From: "VOIP2" <sip:[email protected]>;tag=truoz6lvpq

To: <sip:[email protected];user=phone>;tag=as601fa089

Call-ID: 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9

CSeq: 1 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Max-Forwards: 70

Contact: <sip:[email protected]>

Proxy-Authenticate: Digest realm="voip.fh-stpoelten.ac.at", nonce="4853f723"

Content-Length: 0




---
Scheduling destruction of call '3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9' in 15000 ms
Found user '1002'

voip*CLI> 

<-- SIP read from 192.168.1.7:5060: 
ACK sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-q8684bcfh27w;rport

From: "VOIP2" <sip:[email protected]>;tag=truoz6lvpq

To: <sip:[email protected];user=phone>;tag=as601fa089

Call-ID: 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:[email protected]:5060;line=ge0d7jnc>;flow-id=1

Content-Length: 0





--- (9 headers 0 lines)---

voip*CLI> 

<-- SIP read from 192.168.1.7:5060: 
INVITE sip:[email protected];user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-3illbs1dealc;rport

From: "VOIP2" <sip:[email protected]>;tag=truoz6lvpq

To: <sip:[email protected];user=phone>

Call-ID: 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9

CSeq: 2 INVITE

Max-Forwards: 70

Contact: <sip:[email protected]:5060;line=ge0d7jnc>;flow-id=1

P-Key-Flags: keys="3"

User-Agent: snom320/5.0

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer

Supported: timer, 100rel, replaces, callerid

Session-Expires: 3600;refresher=uas

Proxy-Authorization: Digest username="1002",realm="voip.fh-stpoelten.ac.at",nonce="4853f723",uri="sip:[email protected];user=phone",response="9e0b4d3e7d2c596f1347557e7c8fbcf3",algorithm=md5

Content-Type: application/sdp

Content-Length: 366



v=0

o=root 1072681106 1072681106 IN IP4 192.168.1.7

s=call

c=IN IP4 192.168.1.7

t=0 0

m=audio 10740 RTP/AVP 0 8 9 2 3 18 4 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:2 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:18 g729/8000

a=rtpmap:4 g723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=sendrecv



--- (18 headers 17 lines)---
Using INVITE request as basis request - 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9
Sending to 192.168.1.7 : 5060 (NAT)
Jan 18 08:51:55 WARNING[12241]: acl.c:182 ast_append_ha: NULL is not a valid IP
Jan 18 08:51:55 WARNING[12241]: acl.c:182 ast_append_ha: NULL is not a valid IP
Found user '1002'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 9
Found RTP audio format 2
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.7:10740
Found description format pcmu
Found description format pcma
Found description format g722
Found description format g726-32
Found description format gsm
Found description format g729
Found description format g723
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 85 in default (domain 192.168.1.1)
list_route: hop: <sip:[email protected]:5060;line=ge0d7jnc>
Transmitting (NAT) to 192.168.1.7:5060:
SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-3illbs1dealc;received=192.168.1.7;rport=5060

From: "VOIP2" <sip:[email protected]>;tag=truoz6lvpq

To: <sip:[email protected];user=phone>

Call-ID: 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Max-Forwards: 70

Contact: <sip:[email protected]>

Content-Length: 0




---
Jan 18 08:51:55 WARNING[12238]: acl.c:182 ast_append_ha: NULL is not a valid IP
Jan 18 08:51:55 WARNING[12238]: acl.c:182 ast_append_ha: NULL is not a valid IP
    -- SIP Seeding peer from astdb: '1002' at [email protected]:5060 for 3600
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.7:5060:
OPTIONS sip:[email protected]:5060;line=ge0d7jnc SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK66415e4f;rport

From: "asterisk" <sip:[email protected]>;tag=as3c9eb4e6

To: <sip:[email protected]:5060;line=ge0d7jnc>

Contact: <sip:[email protected]>

Call-ID: [email protected]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 18 Jan 2006 07:51:55 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0




---
Destroying call '[email protected]'

voip*CLI> 
    -- Executing Answer("SIP/1002-f57b", "") in new stack

voip*CLI> 
We're at 192.168.1.1 port 13054
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.7:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-3illbs1dealc;received=192.168.1.7;rport=5060

From: "VOIP2" <sip:[email protected]>;tag=truoz6lvpq

To: <sip:[email protected];user=phone>;tag=as71b7f7a8

Call-ID: 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9

CSeq: 2 INVITE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Max-Forwards: 70

Contact: <sip:[email protected]>

Content-Type: application/sdp

Content-Length: 214



v=0

o=root 12346 12346 IN IP4 192.168.1.1

s=session

c=IN IP4 192.168.1.1

t=0 0

m=audio 13054 RTP/AVP 0 101

a=rtpmap:0 PCMU/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=silenceSupp:off - - - -


---
    -- Executing MeetMe("SIP/1002-f57b", "9000|p") in new stack

  == Parsing '/etc/asterisk/meetme.conf': Found
    -- Created MeetMe conference 1023 for conference '9000'
    -- Playing 'conf-getpin' (language 'de')

voip*CLI> 
Jan 18 08:51:55 WARNING[12238]: acl.c:182 ast_append_ha: NULL is not a valid IP
Jan 18 08:51:55 WARNING[12238]: acl.c:182 ast_append_ha: NULL is not a valid IP
    -- SIP Seeding peer from astdb: '1002' at [email protected]:5060 for 3600
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.7:5060:
OPTIONS sip:[email protected]:5060;line=ge0d7jnc SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK5eec7b2d;rport

From: "asterisk" <sip:[email protected]>;tag=as32a95efd

To: <sip:[email protected]:5060;line=ge0d7jnc>

Contact: <sip:[email protected]>

Call-ID: [email protected]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 18 Jan 2006 07:51:55 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0




---
Destroying call '[email protected]'

voip*CLI> 

<-- SIP read from 192.168.1.7:5060: 
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK66415e4f;rport=5060

From: "asterisk" <sip:[email protected]>;tag=as3c9eb4e6

To: <sip:[email protected]:5060;line=ge0d7jnc>

Call-ID: [email protected]

CSeq: 102 OPTIONS

Contact: <sip:[email protected]:5060;line=ge0d7jnc>;flow-id=1

User-Agent: snom320/5.0

Accept-Language: en

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer

Supported: timer, 100rel, replaces, callerid

Content-Length: 0





--- (14 headers 0 lines)---
Destroying call '[email protected]'

voip*CLI> 

<-- SIP read from 192.168.1.7:5060: 
ACK sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-wb7v8l50sdp3;rport

From: "VOIP2" <sip:[email protected]>;tag=truoz6lvpq

To: <sip:[email protected];user=phone>;tag=as71b7f7a8

Call-ID: 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9

CSeq: 2 ACK

Max-Forwards: 70

Contact: <sip:[email protected]:5060;line=ge0d7jnc>;flow-id=1

Content-Length: 0





--- (9 headers 0 lines)---

voip*CLI> 

<-- SIP read from 192.168.1.7:5060: 
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK5eec7b2d;rport=5060

From: "asterisk" <sip:[email protected]>;tag=as32a95efd

To: <sip:[email protected]:5060;line=ge0d7jnc>

Call-ID: [email protected]

CSeq: 102 OPTIONS

Contact: <sip:[email protected]:5060;line=ge0d7jnc>;flow-id=1

User-Agent: snom320/5.0

Accept-Language: en

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer

Supported: timer, 100rel, replaces, callerid

Content-Length: 0





--- (14 headers 0 lines)---
Destroying call '[email protected]'

voip*CLI> 

<-- SIP read from 192.168.1.7:5060: 
BYE sip:[email protected] SIP/2.0

Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-p44bvkze549h;rport

From: "VOIP2" <sip:[email protected]>;tag=truoz6lvpq

To: <sip:[email protected];user=phone>;tag=as71b7f7a8

Call-ID: 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9

CSeq: 3 BYE

Max-Forwards: 70

Contact: <sip:[email protected]:5060;line=ge0d7jnc>;flow-id=1

User-Agent: snom320/5.0

RTP-RxStat: Total_Rx_Pkts=133,Rx_Pkts=133,Rx_Pkts_Lost=0,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=122,Tx_Pkts=122,Remote_Tx_Pkts=0

Content-Length: 0





--- (12 headers 0 lines)---
Sending to 192.168.1.7 : 5060 (NAT)
Transmitting (NAT) to 192.168.1.7:5060:
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.7:5060;branch=z9hG4bK-p44bvkze549h;received=192.168.1.7;rport=5060

From: "VOIP2" <sip:[email protected]>;tag=truoz6lvpq

To: <sip:[email protected];user=phone>;tag=as71b7f7a8

Call-ID: 3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9

CSeq: 3 BYE

User-Agent: Asterisk PBX

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Max-Forwards: 70

Contact: <sip:[email protected]>

Content-Length: 0




---

voip*CLI> 
    -- Hungup 'Zap/pseudo-1620342069'
  == Spawn extension (default, 85, 2) exited non-zero on 'SIP/1002-f57b'

voip*CLI> 
Jan 18 08:51:58 WARNING[12238]: acl.c:182 ast_append_ha: NULL is not a valid IP
Jan 18 08:51:58 WARNING[12238]: acl.c:182 ast_append_ha: NULL is not a valid IP
    -- SIP Seeding peer from astdb: '1002' at [email protected]:5060 for 3600
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.7:5060:
OPTIONS sip:[email protected]:5060;line=ge0d7jnc SIP/2.0

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK63b7969d;rport

From: "asterisk" <sip:[email protected]>;tag=as24e68450

To: <sip:[email protected]:5060;line=ge0d7jnc>

Contact: <sip:[email protected]>

Call-ID: [email protected]

CSeq: 102 OPTIONS

User-Agent: Asterisk PBX

Max-Forwards: 70

Date: Wed, 18 Jan 2006 07:51:58 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

Content-Length: 0




---
Destroying call '[email protected]'

voip*CLI> 

<-- SIP read from 192.168.1.7:5060: 
SIP/2.0 200 OK

Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK63b7969d;rport=5060

From: "asterisk" <sip:[email protected]>;tag=as24e68450

To: <sip:[email protected]:5060;line=ge0d7jnc>

Call-ID: [email protected]

CSeq: 102 OPTIONS

Contact: <sip:[email protected]:5060;line=ge0d7jnc>;flow-id=1

User-Agent: snom320/5.0

Accept-Language: en

Accept: application/sdp

Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO

Allow-Events: talk, hold, refer

Supported: timer, 100rel, replaces, callerid

Content-Length: 0





--- (14 headers 0 lines)---
Destroying call '[email protected]'
Destroying call '3c290d23493e-q8u1b5xt7wi2@snom320-0004132413E9'

voip*CLI> 

voip*CLI> 

voip*CLI> 

voip*CLI>

das war einfach ein call zu einem meetme konferenzraum (angerufen, die ansage angehört, aufgelegt).

wenn ich das selbe mit einem cisco phone mache, dann bekomme ich den fehler nicht.

irgendwelche ideen?

mfg,
michael
 
Kannst du mal bitte noch deine sip.conf posten?
 
das ganze ist realtime

sip.conf

Code: 
[general]
context=default         ; Default context for incoming calls
allowguest=no                   ; Allow or reject guest calls (default is yes, this can also be set to 'osp'
                                ; if asterisk was compiled with OSP support.
realm=voip.fh-stpoelten.ac.at   ; Realm for digest authentication
                                ; defaults to "asterisk"
                                ; Realms MUST be globally unique according to RFC 3261
                                ; Set this to your host name or domain name
bindport=5060                   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound calls
                                ; Note: Asterisk only uses the first host
                                ; in SRV records
                                ; Disabling DNS SRV lookups disables the
                                ; ability to place SIP calls based on domain
                                ; names to some other SIP users on the Internet

;domain=mydomain.tld            ; Set default domain for this host
                                ; If configured, Asterisk will only allow
                                ; INVITE and REFER to non-local domains
                                ; Use "sip show domains" to list local domains
;domain=mydomain.tld,mydomain-incoming
                                ; Add domain and configure incoming context
                                ; for external calls to this domain
;domain=1.2.3.4                 ; Add IP address as local domain
                                ; You can have several "domain" settings
;tos=184                        ; Set IP QoS to either a keyword or numeric val
;tos=lowdelay                   ; lowdelay,throughput,reliability,mincost,none

disallow=all                    ; First disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
                                ; This may also be set for individual users/peers
language=de                     ; Default language setting for all users/peers
                                ; This may also be set for individual users/peers

dtmfmode=inband

canreinvite=no

und in der sip_users:
Code: 
 vollständige Textfelder  	
id   name   accountcode   amaflags   callgroup   callerid   canreinvite   context   defaultip   dtmfmode   fromuser   fromdomain   fullcontact   host   insecure   language   mailbox   md5secret   nat   deny   permit   mask   pickupgroup   port   qualify   restrictcid   rtptimeout   rtpholdtimeout   secret   type   username   disallow   allow   musiconhold   regseconds   ipaddr   regexten   cancallforward
1 	1000 	NULL 	NULL 	NULL 	Michael Prochaska <1000> 	no 	default 	NULL 	rfc2833 	NULL 	NULL 	NULL 	dynamic 	NULL 	de 	1000@default 	NULL 	yes 	NULL 	NULL 	NULL 	NULL 	`7 	yes 	NULL 	NULL 	NULL 	asti02 	friend 	1000 	all 	ulaw 	NULL 	0 	p7 	  	yes
2 	1001 	NULL 	NULL 	NULL 	eCampus <1001> 	no 	default 	NULL 	inband 	NULL 	NULL 	NULL 	dynamic 	NULL 	de 	1001@default 	NULL 	no 	NULL 	NULL 	NULL 	NULL 	5060 	no 	NULL 	NULL 	NULL 	atwi02 	friend 	1001 	all 	ulaw 	NULL 	1137591208 	192.168.1.90 	  	yes
3 	1002 	NULL 	NULL 	NULL 	Michael Prochaska <1002> 	no 	default 	NULL 	rfc2833 	NULL 	NULL 	NULL 	dynamic 	NULL 	de 	1002@default 	NULL 	yes 	NULL 	NULL 	NULL 	NULL 	5060 	yes 	NULL 	NULL 	NULL 	asti02 	friend 	1002 	all 	ulaw 	NULL 	1137593363 	192.168.1.7 	  	yes

NAT und qualify hab ich schon geändert, hat auch nix gebracht.

mfg,
michael
 
Das scheint ein Fehler beim parsen der permit/deny-Werte zu sein. Versuch mal da einen leeren String zu setzen statt NULL.
 
hey cool, das wars.

wäre es dann nicht sinnvoll, bei allen optionen den defaultwert auf "" anstatt NULL zu setzen?

mfg,
michael
 
Das kannst du ja mal testen aber es könnte auch passieren, dass dadurch wieder neue Probleme entstehen (obwohl ich das jetzt eher nicht vermute).
 
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Die Verbindung, Zusammenarbeit und Kommunikation mit Ihrem Team und Ihren Kunden war noch nie so einfach. Nutzen Sie unsere kostenlose 30-Tage-Testversion und entdecken Sie die richtige Lösung für Ihr Unternehmen.

3CX start your free trial guy
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