Echec d'enregistrement telephone SIP

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18/11/2010, 12h33
chab

Echec d'enregistrement telephone SIP

Bonjour,

j'ai un probleme et je ne sais plus quoi faire, j'espere que vous pourrez m'aider :

J'ai un telephone SIP Siemens C470IP qui marche bien avec divers providers VoIP (OVH, Betamax...). Il marchait bien aussi avec ma machine de test sous ubuntu et en reseau local.

Le probleme est que maintenant le telephone ne se connecte plus sur le serveur dédié que j'ai (sous debian lenny / Asterisk 1.6.2.13). Par contre Ekiga se connecte sans probleme au meme compte sip...

Au niveau de la console asterisk (meme en tres verbeux), je n'ai rien d'affiché lorsque l'enregistrement echoue et le C470 me dit juste "Registration failed". Sans plus de logs difficile d'avancer :(

voici un extrait de mon sip.conf :

Code:

[general] dtmfmode=auto language=fr ; pour les messages lus par asterisk disallow=all allow=ulaw allow=alaw allow=speex [siemens] type=friend context=interne host=dynamic secret=xxxxxxxx

Si vous avez une suggestion de test que je puisse faire parce que la j'ai pas d'idée...
18/11/2010, 13h48
ds3

Bonjour,
Le soft phone qui s'enregistre et pas le C470 avec le meme compte SIP client.
Pas de bla bla de la console ... même avec mode verbeux maximal ...

Adresse ip du C470 ? sip set debug ip XXX.XXX.XXX.XXX
ou un tcpdump tcpdump -s 0 -i ethX
... une idée rapide.
18/11/2010, 14h07
chab

Merci j'obtiens enfin quelque chose avec "sip set debug ip X.X.X.X"... Apres ca m'avance pas enormement pour le moment.
Je vais chercher de mon coté mais si quelque chose vous saute au yeux, voici les logs :

Code:

Debian-50-lenny-32-minimal*CLI> sip set debug ip 89.4.1xx.2xx SIP Debugging Enabled for IP: 89.4.1xx.2xx Debian-50-lenny-32-minimal*CLI> Debian-50-lenny-32-minimal*CLI> Debian-50-lenny-32-minimal*CLI> <--- SIP read from UDP:89.4.1xx.2xx:61177 ---> <-------------> <--- SIP read from UDP:89.4.1xx.2xx:61177 ---> REGISTER sip:server.exemple.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK375a1a8b1ad99205b5e56896d094e0f1;rport From: "siemens" <sip:siemens@server.exemple.com>;tag=661213739 To: "siemens" <sip:siemens@server.exemple.com> Call-ID: 994182810@192_168_0_50 CSeq: 1 REGISTER Contact: <sip:siemens@192.168.0.50:5060> Max-Forwards: 70 User-Agent: C470IP022230000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:89.4.1xx.2xx:61177 ---> REGISTER sip:server.exemple.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK375a1a8b1ad99205b5e56896d094e0f1;rport From: "siemens" <sip:siemens@server.exemple.com>;tag=661213739 To: "siemens" <sip:siemens@server.exemple.com> Call-ID: 994182810@192_168_0_50 CSeq: 1 REGISTER Contact: <sip:siemens@192.168.0.50:5060> Max-Forwards: 70 User-Agent: C470IP022230000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- [Nov 18 13:03:30] NOTICE[17316]: chan_sip.c:11601 sip_reregister: -- Re-registration for 003397453xxxx@sip.ovh.net > doing dnsmgr_lookup for 'sip.ovh.net' > doing dnsmgr_lookup for 'sip.ovh.net' [Nov 18 13:03:30] NOTICE[17316]: chan_sip.c:18334 handle_response_register: Outbound Registration: Expiry for sip.ovh.net is 120 sec (Scheduling reregistration in 105 s) <--- SIP read from UDP:89.4.1xx.2xx:61177 ---> REGISTER sip:server.exemple.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK375a1a8b1ad99205b5e56896d094e0f1;rport From: "siemens" <sip:siemens@server.exemple.com>;tag=661213739 To: "siemens" <sip:siemens@server.exemple.com> Call-ID: 994182810@192_168_0_50 CSeq: 1 REGISTER Contact: <sip:siemens@192.168.0.50:5060> Max-Forwards: 70 User-Agent: C470IP022230000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:89.4.1xx.2xx:61177 ---> REGISTER sip:server.exemple.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK375a1a8b1ad99205b5e56896d094e0f1;rport From: "siemens" <sip:siemens@server.exemple.com>;tag=661213739 To: "siemens" <sip:siemens@server.exemple.com> Call-ID: 994182810@192_168_0_50 CSeq: 1 REGISTER Contact: <sip:siemens@192.168.0.50:5060> Max-Forwards: 70 User-Agent: C470IP022230000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:89.4.1xx.2xx:61177 ---> REGISTER sip:server.exemple.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK375a1a8b1ad99205b5e56896d094e0f1;rport From: "siemens" <sip:siemens@server.exemple.com>;tag=661213739 To: "siemens" <sip:siemens@server.exemple.com> Call-ID: 994182810@192_168_0_50 CSeq: 1 REGISTER Contact: <sip:siemens@192.168.0.50:5060> Max-Forwards: 70 User-Agent: C470IP022230000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:89.4.1xx.2xx:61177 ---> REGISTER sip:server.exemple.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK375a1a8b1ad99205b5e56896d094e0f1;rport From: "siemens" <sip:siemens@server.exemple.com>;tag=661213739 To: "siemens" <sip:siemens@server.exemple.com> Call-ID: 994182810@192_168_0_50 CSeq: 1 REGISTER Contact: <sip:siemens@192.168.0.50:5060> Max-Forwards: 70 User-Agent: C470IP022230000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- <--- SIP read from UDP:89.4.1xx.2xx:61177 ---> REGISTER sip:server.exemple.com SIP/2.0 Via: SIP/2.0/UDP 192.168.0.50:5060;branch=z9hG4bK375a1a8b1ad99205b5e56896d094e0f1;rport From: "siemens" <sip:siemens@server.exemple.com>;tag=661213739 To: "siemens" <sip:siemens@server.exemple.com> Call-ID: 994182810@192_168_0_50 CSeq: 1 REGISTER Contact: <sip:siemens@192.168.0.50:5060> Max-Forwards: 70 User-Agent: C470IP022230000000 Expires: 180 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Debian-50-lenny-32-minimal*CLI>
18/11/2010, 15h17
ds3

Tu as modifié quelques choses de ces INVITE's ?
Ou une info pertinente de /etc/hosts ?
18/11/2010, 15h31
chab

je suis un utilisateur debutant d'asterisk et je n'ai rien modifié de particulier, ni sur telephone ni sur asterisk.

mon /etc/host n'a rien de spécial :

Code:

# nameserver config # IPv4 127.0.0.1 localhost 78.47.xxx.xxx Debian-50-lenny-32-minimal # # IPv6 ::1 ip6-localhost ip6-loopback fe00::0 ip6-localnet ff00::0 ip6-mcastprefix ff02::1 ip6-allnodes ff02::2 ip6-allrouters ff02::3 ip6-allhosts

petit parenthese sur le serveur dédié, c'est un VPS en allemagne chez hetner:
http://www.hetzner.de/en/hosting/produkte_vserver/vq7/
il ne coute que 8€ ttc par mois et depuis 15 jours il marche très bien, sauf pour asterisk :)
18/11/2010, 15h54
chab

etant débutant dans le domaine d'asterisk, je n'ai rien modifié de spécial...

rien de particulier non plus dans le /etc/hosts.

Pour info voici les logs lorsque je connecte ekiga avec le même compte (siemens) et depuis la même connection internet. On y voit clairement que asterisk repond gentillement a ekiga alors qu avec le telephone asterisk ne repond rien....

Code:

<--- SIP read from UDP:89.4.2xx.1xx:61772 ---> REGISTER sip:server.domain.com SIP/2.0 CSeq: 5 REGISTER Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKd71ac623-c502-1910-9d9a-000f66cf7cc5;rport User-Agent: Ekiga/3.2.7 From: <sip:siemens@server.domain.com>;tag=b1c8c523-c502-1910-9d9a-000f66cf7cc5 Call-ID: b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi To: <sip:siemens@server.domain.com> Contact: <sip:siemens@89.4.2xx.1xx:61772>;q=1, <sip:siemens@89.4.2xx.1xx>;q=0.667, <sip:siemens@192.168.0.10>;q=0.334 llow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 <-------------> --- (12 headers 0 lines) --- Sending to 89.4.2xx.1xx : 61772 (no NAT) <--- Transmitting (no NAT) to 89.4.2xx.1xx:61772 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKd71ac623-c502-1910-9d9a-000f66cf7cc5;received=89.4.2xx.1xx;rport=61772 From: <sip:siemens@server.domain.com>;tag=b1c8c523-c502-1910-9d9a-000f66cf7cc5 To: <sip:siemens@server.domain.com>;tag=as3384ec04 Call-ID: b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi CSeq: 5 REGISTER Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78df32a9" Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:89.4.2xx.1xx:61772 ---> REGISTER sip:server.domain.com SIP/2.0 CSeq: 6 REGISTER Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKdb2ec823-c502-1910-9d9a-000f66cf7cc5;rport User-Agent: Ekiga/3.2.7 Authorization: Digest username="siemens", realm="asterisk", nonce="78df32a9", uri="sip:server.domain.com", algorithm=MD5, response="fec67185c669f9a966e67d5194284a93" From: <sip:siemens@server.domain.com>;tag=b1c8c523-c502-1910-9d9a-000f66cf7cc5 Call-ID: b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi To: <sip:siemens@server.domain.com> Contact: <sip:siemens@89.4.2xx.1xx:61772>;q=1, <sip:siemens@89.4.2xx.1xx>;q=0.667, <sip:siemens@192.168.0.10>;q=0.334 Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Content-Length: 0 Max-Forwards: 70 <-------------> --- (13 headers 0 lines) --- Sending to 89.4.2xx.1xx : 61772 (no NAT) -- Registered SIP 'siemens' at 89.4.2xx.1xx port 61772 > Saved useragent "Ekiga/3.2.7" for peer siemens <--- Transmitting (no NAT) to 89.4.2xx.1xx:61772 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKdb2ec823-c502-1910-9d9a-000f66cf7cc5;received=89.4.2xx.1xx;rport=61772 From: <sip:siemens@server.domain.com>;tag=b1c8c523-c502-1910-9d9a-000f66cf7cc5 To: <sip:siemens@server.domain.com>;tag=as3384ec04 Call-ID: b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi CSeq: 6 REGISTER Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Expires: 3600 Contact: <sip:siemens@89.4.2xx.1xx:61772>;expires=3600 Date: Thu, 18 Nov 2010 13:44:25 GMT Content-Length: 0 <------------> Scheduling destruction of SIP dialog 'b1c8c523-c502-1910-9d99-000f66cf7cc5@cdi' in 32000 ms (Method: REGISTER) <--- SIP read from UDP:89.4.2xx.1xx:61772 ---> SUBSCRIBE sip:siemens@server.domain.com SIP/2.0 CSeq: 2 SUBSCRIBE Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bK4ca0c823-c502-1910-9d9a-000f66cf7cc5;rport User-Agent: Ekiga/3.2.7 From: <sip:siemens@server.domain.com>;tag=639cc823-c502-1910-9d9a-000f66cf7cc5 Call-ID: 7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi To: <sip:siemens@server.domain.com> Contact: <sip:siemens@89.4.2xx.1xx:61772> Accept: application/simple-message-summary Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Event: message-summary Content-Length: 0 Max-Forwards: 70 <-------------> --- (14 headers 0 lines) --- Creating new subscription Sending to 89.4.2xx.1xx : 61772 (no NAT) list_route: hop: <sip:siemens@89.4.2xx.1xx:61772> Found peer 'siemens' for 'siemens' from 89.4.2xx.1xx:61772 <--- Transmitting (no NAT) to 89.4.2xx.1xx:61772 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bK4ca0c823-c502-1910-9d9a-000f66cf7cc5;received=89.4.2xx.1xx;rport=61772 From: <sip:siemens@server.domain.com>;tag=639cc823-c502-1910-9d9a-000f66cf7cc5 To: <sip:siemens@server.domain.com>;tag=as0f9d2526 Call-ID: 7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi CSeq: 2 SUBSCRIBE Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33349611" Content-Length: 0 <------------> Scheduling destruction of SIP dialog '7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi' in 32000 ms (Method: SUBSCRIBE) <--- SIP read from UDP:89.4.2xx.1xx:61772 ---> SUBSCRIBE sip:siemens@server.domain.com SIP/2.0 CSeq: 3 SUBSCRIBE Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKbe11c923-c502-1910-9d9a-000f66cf7cc5;rport User-Agent: Ekiga/3.2.7 Authorization: Digest username="siemens", realm="asterisk", nonce="33349611", uri="sip:siemens@server.domain.com", algorithm=MD5, response="4aa386369a65250b3c10a494a5b42da9" From: <sip:siemens@server.domain.com>;tag=639cc823-c502-1910-9d9a-000f66cf7cc5 Call-ID: 7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi To: <sip:siemens@server.domain.com> Contact: <sip:siemens@89.4.2xx.1xx:61772> Accept: application/simple-message-summary Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING Expires: 3600 Event: message-summary Content-Length: 0 Max-Forwards: 70 <-------------> --- (15 headers 0 lines) --- Creating new subscription Sending to 89.4.2xx.1xx : 61772 (no NAT) Found peer 'siemens' for 'siemens' from 89.4.2xx.1xx:61772 <--- Transmitting (no NAT) to 89.4.2xx.1xx:61772 ---> SIP/2.0 404 Not found (no mailbox) Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bKbe11c923-c502-1910-9d9a-000f66cf7cc5;received=89.4.2xx.1xx;rport=61772 From: <sip:siemens@server.domain.com>;tag=639cc823-c502-1910-9d9a-000f66cf7cc5 To: <sip:siemens@server.domain.com>;tag=as0f9d2526 Call-ID: 7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi CSeq: 3 SUBSCRIBE Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> [Nov 18 14:44:26] NOTICE[17316]: chan_sip.c:21518 handle_request_subscribe: Received SIP subscribe for peer without mailbox: siemens Really destroying SIP dialog '7a98c823-c502-1910-9d9a-000f66cf7cc5@cdi' Method: SUBSCRIBE <--- SIP read from UDP:89.4.2xx.1xx:61772 ---> PUBLISH sip:siemens@server.domain.com SIP/2.0 CSeq: 7 PUBLISH Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bK81c3c823-c502-1910-9d9a-000f66cf7cc5;rport User-Agent: Ekiga/3.2.7 From: <sip:siemens@server.domain.com>;tag=afbbc823-c502-1910-9d9a-000f66cf7cc5 Call-ID: dcb3c823-c502-1910-9d9a-000f66cf7cc5@cdi To: <sip:siemens@server.domain.com> Contact: <sip:siemens@89.4.2xx.1xx:61772> Expires: 500 Event: presence Content-Type: application/pidf+xml Content-Length: 346 Max-Forwards: 70 <?xml version="1.0" encoding="UTF-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" entity="pres:siemens@server.domain.com"> <tuple id="sip:siemens@server.domain.com_on_cdi"> <note>online - I'm online using Ekiga</note> <status> <basic>open</basic> </status> <contact priority="1">siemens@server.domain.com</contact> </tuple> </presence> <-------------> --- (13 headers 10 lines) --- <--- Transmitting (no NAT) to 89.4.2xx.1xx:61772 ---> SIP/2.0 501 Method Not Implemented Via: SIP/2.0/UDP 89.4.2xx.1xx:61772;branch=z9hG4bK81c3c823-c502-1910-9d9a-000f66cf7cc5;received=89.4.2xx.1xx;rport=61772 From: <sip:siemens@server.domain.com>;tag=afbbc823-c502-1910-9d9a-000f66cf7cc5 To: <sip:siemens@server.domain.com>;tag=as62cfd18d Call-ID: dcb3c823-c502-1910-9d9a-000f66cf7cc5@cdi CSeq: 7 PUBLISH Server: Asterisk PBX 1.6.2.13 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 <------------> <--- SIP read from UDP:89.4.2xx.1xx:61772 ---> <-------------> Debian-50-lenny-32-minimal*CLI>
19/11/2010, 15h30
chab

Ok en fait c'etait juste un probleme NAT... il suffisait donc de mettre un serveur stun dans la config du tel.

Comme on peut le voir dans les logs ci dessus, l'adresse de la requete envoyée par le telephone etait l'adresse locale (192.168.0.50) d'ou le probleme.
Ekiga quant à lui donne bien l'adresse publique dans l'invite...

Ce site explique tres bien tout cela avec des petits schemas et tout:
http://www.ordinoscope.net/index.php...tes/SIP_et_Nat