snom Canada

As a workaround please use co-lines. create trunk for inbound calls co-lines : co1 co2 co3 co4.... co19 send call to extension : ACD extension. direction : inbound This way you would be limiting the number of calls in. if the number of calls is larger than the coXX it will be busy signal. create trunk for outbound calls nothing special is needed for this trunk except direction : outbound. in ACD leave maximum calls to unlimited. (this will be change in next release.) The other way will be to add i or o after co1 = co1:i co2:o co3 if you use only one trunk. co3 will inbound and outbound. version 4.3.0.5021 should be used. (for production system)

Bonjour, voici la marche à suivre pour activer un trunk avec telephonic.ca Aller dans trunk crée le trunk : telephonic Cliquer sur le trunk. Cliquer sur « here » to switch in text mode. Les valeurs soulignées sont à changer. ATTENTION : Activer votre pbx/license dans le tab license. Coller le texte suivant : Name: telephonic.ca Type: gateway To: sip RegPass: ******** Direction: Disabled: false Global: false Display: RegAccount: RegRegistrar: telephonic.ca RegKeep: RegUser: Icid: Require: OutboundProxy: sip:sip.telephonic.ca:5060;transport=udp Ani: 1514NXX6162 DialExtension: 100 Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: never Privacy: false Glob: noplus RequestTimeout: Codecs: CodecLock: true Expires: 3600 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: 69.28.215.131 216.187.103.190 InterOffice: false DialPlan: UseEpid: false Ignore18xSDP: false Colines: DialogPermission:

Bonjour, voici la marche à suivre pour activer un trunk avec broadconnect.ca Aller dans trunk crée le trunk : broadconnect Cliquer sur le trunk. Cliquer sur « here » to switch in text mode. Les valeurs soulignées sont à changer. ATTENTION : Activer votre pbx/license dans le tab license. Coller le texte suivant : Name: broadconnect Type: register To: sip RegPass: ******** Direction: Disabled: false Global: false Display: 514NXX3456 RegAccount: 514NXX3456 RegRegistrar: broadconnect.ca RegKeep: 60 RegUser: 514NXX3456 Icid: Require: OutboundProxy: sip:sip.broadconnect.ca:5060;transport=udp Ani: 514NXX3456 DialExtension: 100 Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: never Privacy: rpi Glob: RequestTimeout: Codecs: 0 CodecLock: true Expires: 120 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: UseEpid: false Ignore18xSDP: false Colines: DialogPermission:

Ok, il semble que une chose soit regler. (la license) J'ai vu que des fois ca ne marche pas.. erreur 408. On va travailler avec Modulis pour voir comment ameliorer cela. Que nous reste-t-il a faire ? pour un revois d'appel. il y a des actions optionelles 3,4 (si tu veux une receptioniste automatisée) 1. create trunk (done) 2. dans trunk send call to extention XYZ 3. create auto attendant XYZ 4. dans AA direct input option 1 destination XXY 5. create extenstion xxy 6. modifier dans extension > redirection > cell number XXXYYYZZZZ 7. modifier dans extension > redirection > Include the cell phone in calls to extension: immediate

click trunk en haut pour refresh la page. Si tu as les log je vais regarder. (nous avons aussi des 408 request timeout des fois 1 par heure 1 sur 60 request) Mettre: sip logging > call message = yes logwatch level 8 tous les autres a no De notre cote [5] 2011/11/18 05:23:53: Registration on trunk 1 (modulis) failed. Retry in 60 seconds [2] 2011/11/18 05:23:53: Trunk status modulis (1) changed to "408 Request Timeout" (Registration failed, retry after 60 seconds) [8] 2011/11/18 05:24:53: Trunk 1: Preparing for re-registration [8] 2011/11/18 05:24:53: Trunk 1: sending discover message for 72.55.134.154 [8] 2011/11/18 05:24:53: Trunk 1: Received reply for discover method [8] 2011/11/18 05:24:53: Trunk 1 (modulis) is associated with the following addresses: udp:72.55.134.154:5060 [8] 2011/11/18 05:24:53: Trunk modulis: Sending registration to sip:72.55.134.154:5060;transport=udp [8] 2011/11/18 05:24:53: Answer challenge with username 1158 [8] 2011/11/18 05:24:53: SIP Rx udp:72.55.134.154:5060: OPTIONS sip:1158@192.168.1.99:5060;transport=udp;line=c4ca4238 SIP/2.0 Via: SIP/2.0/UDP 72.55.134.154:5060;branch=z9hG4bK7423f576;rport From: "asterisk" <sip:asterisk@72.55.134.154:5060>;tag=as0c63046a To: <sip:1158@192.168.1.99:5060;transport=udp;line=c4ca4238> Contact: <sip:asterisk@72.55.134.154:5060> Call-ID: 78455be35d0821a96e3d0e3b71f19ae8@72.55.134.154 CSeq: 102 OPTIONS User-Agent: MODULIS-GW01 Max-Forwards: 70 Date: Fri, 18 Nov 2011 10:24:48 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 [8] 2011/11/18 05:24:53: SIP Tx udp:72.55.134.154:5060: SIP/2.0 200 Ok Via: SIP/2.0/UDP 72.55.134.154:5060;branch=z9hG4bK7423f576;rport=5060 From: "asterisk" <sip:asterisk@72.55.134.154:5060>;tag=as0c63046a To: <sip:1158@192.168.1.99:5060;transport=udp;line=c4ca4238>;tag=81232eb1c8 Call-ID: 78455be35d0821a96e3d0e3b71f19ae8@72.55.134.154 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Content-Length: 0 [2] 2011/11/18 05:24:53: Trunk status modulis (1) changed to "200 OK" (Refresh interval 60 seconds) [8] 2011/11/18 05:24:53: Trunk 1: setup callback to send re-registration after 60 seconds

Ca semble mieux, Tu devrais etre capable de register. click sur le lien REGISTER en bleu a cote du trunk modulis si tu as toujours le 408. assure toi d'avoir que le pbx et ton mac sur le réseau. Pas d'autre switch ou router.

Qui est ton provider internet. Quel pays? allemangne? Ce tracert n'est vraiment pas bon... Tu n'arivera a rien avec un lien de cette qualité. Tu devrais avoir quelque chose comme... Si tu es au Québec. Pinging gw01.modulis.ca [72.55.134.154] with 32 bytes of data: Reply from 72.55.134.154: bytes=32 time=11ms TTL=58 Reply from 72.55.134.154: bytes=32 time=14ms TTL=58 Reply from 72.55.134.154: bytes=32 time=10ms TTL=58 Reply from 72.55.134.154: bytes=32 time=16ms TTL=58 Tracing route to gw01.modulis.ca [72.55.134.154] over a maximum of 30 hops: 1 * 3 ms 3 ms 192.168.1.1 2 12 ms 19 ms 10 ms 10.86.168.1 3 29 ms 16 ms 25 ms 10.170.172.229 4 21 ms 20 ms 12 ms 10.170.171.110 5 26 ms 11 ms 15 ms 216.113.123.121 6 12 ms 12 ms 11 ms videotron-gw.peer1.net [216.187.115.253] 7 13 ms 11 ms 12 ms 216.187.120.169 8 10 ms 12 ms 9 ms te6-3.cl-core05.peer1.mtl.iweb.com [216.187.90.134] 9 12 ms 12 ms 12 ms 72-55-134-154.modulis.ca [72.55.134.154] Trace complete.

mettre le ip au lieu du nom domain... c'est un probleme de vitesse pour trouver le site. erreur 408 c'est un delais. aussi faire un ping sur le gw01.modulis.ca le delais doit etre en bas de 20ms. S'il y a des ratés ce n'est pas bon non plus. peut-etre aussi vérifier le trace route combien de hop entre les deux?

Dans le router nbg4604 Network > NAT > Application service name : rtp local port range : 49152 ~ 65535 public port range : 49152 ~ 65535 protocol : tcp/udp server IP : pbx ip 192.168.x.y service name : sip local port range : 5060 ~ 5061 public port range : 5060 ~ 5061 protocol : tcp/udp server IP : pbx ip 192.168.x.y Management > Bandwidth MGMT > General Bandwidth Management Type : Priority queue mettre les bonnes valeurs selon le lien internet Management > Bandwidth MGMT > Advanced activer 1,2,3,4 low de-activer 5 6 7 ajouter 8 similaire a rtp high ajouter 9 similaire a sip high

Bonjour, voici la marche à suivre pour activer un trunk avec modulis.ca Aller dans trunk crée le trunk : modulis Cliquer sur le trunk Cliquer sur « here » to switch in text mode Les valeurs soulignées sont à changer utiliser le ip 72.55.134.154 au lieu du gw01.modulis.ca (408 = delais/erreurs dns) soyer sur d'activer votre pbx/license dans le tab license. Coller le texte suivant : # Trunk 1 in domain pbx.abc.com Name: modulis Type: register To: sip RegPass: ******** Direction: Disabled: false Global: false Display: RegAccount: 1158 RegRegistrar: 72.55.134.154 RegKeep: 30 RegUser: 1158 Icid: Require: OutboundProxy: sip:72.55.134.154:5060;transport=udp Ani: VOTRE DID 514xxxYYYY DialExtension: 100 Prefix: Trusted: false AcceptRedirect: false RfcRtp: false Analog: false SendEmail: UseUuid: false Ring180: false Failover: never Privacy: pai Glob: RequestTimeout: Codecs: 0 CodecLock: true Expires: 60 FromUser: Tel: true TranscodeDtmf: false AssociatedAddresses: InterOffice: false DialPlan: UseEpid: false Ignore18xSDP: false Colines: DialogPermission:

we do not have white noise issue, we have no audio/nothing on all call we do. call we receive the audio is back/ok. pbxnsip 1.5.2.10a on windows2000 server (running for more than 18 months.) broken firmware : 6.5.13 6.5.15 working firmware : 6.5.8 6.5.9 beta 6.5.10 (snom320) 7.1.30 (snom 360)

Hello, Warning, a major issue is ATTACKING all are phone at this moment. 6.5.15 update on a Snom 360 has no voice when you make the call. When receiving the call everything ok. PBXNSIP vewrsion 1.5.2.10a downgrade to 6.5.8 now ok... but white noise might come back. 6.5.10 ok also automatic update did scrap all the phone we have. need 6.5.12 and 6.5.10 firmware (snom provide only 6.5.13 and 6.5.15) URGENT solution is needed. no other phone has it (GRANDSTREAM, AASTRA) regards,

Hello, Does anyone have installed Cisco 7941/7961 FW 8.0.3 with pbxnsip 1.5.2.10a or 2.x? If you do please let me know. Does anyone have these two files mk-sip.jar and 3g-tones.xml for the cisco phone 7941/7961. Did you intall behind firewall/router and NAT Would need the SEP{mac}.cnf.xml sample files to validated if we are ok. best regards, Digisoft VoIP

We will try to capture the traffic. The equipment seems to be find we have many installations unsing these ZyXEL router. Not all in voip solution! Also use the NetGear FSM7326P POE switch. We have found that if there is more number to register in the same phone, it would create more problems. Is there any auto reboot when registration fail? DigiVoip

In snom phone UDP is default no TCP used. When you do not write anything transport is udp (similar to transport=udp) IP address of the proxy is used and no DNS are involve. will have to think about something else. regards,

If you have test and know this flag could be please let me know. There is a setting call <voipControlPort> but somehow pbxnsip do not respect that. Even if the request is properly made to pbxnsip. pbxnsip log >>>> Contact: <sip:5149048083@192.168.3.81:5067;transport=udp> pbxnsip try to contact on a diferent port 49157 we do not understand why this port is used. THIS IS NOT OK! ***** After ..... We have done a test, use 49157 in voipControlPort [0] 20071008112635: SIP Tm udp:200.123.47.50:25195: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.3.81:49157;branch=z9hG4bK0b10f88f;rport=25195;received=200.123.47.50 From: <sip:5149048083@66.46.213.1>;tag=0017e03419da00069aa3774f-92fc4a8f To: <sip:5149048083@66.46.213.1>;tag=32250 Call-ID: 0017e034-19da0002-42362800-44b269f6@192.168.3.81 CSeq: 105 REGISTER Contact: <sip:5149048083@192.168.3.81:49157;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0017e03419da>";+u.sip!model.ccm.cisco.com=30018;expires=90 Content-Length: 0 [0] 20071008112636: SIP Rx udp:200.123.47.50:25195: REGISTER sip:66.46.213.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.81:49157;branch=z9hG4bK0b10f88f From: <sip:5149048083@66.46.213.1>;tag=0017e03419da00069aa3774f-92fc4a8f To: <sip:5149048083@66.46.213.1> Call-ID: 0017e034-19da0002-42362800-44b269f6@192.168.3.81 Max-Forwards: 70 Date: Mon, 08 Oct 2007 15:26:34 GMT CSeq: 105 REGISTER User-Agent: Cisco-CP7961G/8.3.0 Contact: <sip:5149048083@192.168.3.81:49157;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0017e03419da>";+u.sip!model.ccm.cisco.com="30018" Supported: (null),X-cisco-xsi-6.0.1 Content-Length: 0 Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP0017E03419DA Load=SIP41.8-3-1S Last=Phone-Keypad" Expires: 3600 PS : Somehow the request came back on 49158 seen in wireshark. ***** explanation : <voipControlPort> UDP port to listen for incoming SIP messages (defaults to 5060). Note that this is not the port the phone uses to send SIP messages. This odd behavior is RFC compliant but highly non-standard and breaks symmetric NAT traversal workarounds commonly deployed by VOIP providers.

For your info : the phone never register back after a fail. Absolute need to reboot... Case 1 : the phone snom 360 has many numbers, only one of them fail. Case 2 : the network has many phone snom 360 and snom 320 only one fail at a given time. Case 3 : the network and the phone have many different type of registration UDP and TLS. Still same issues some will fail some will continue to work. We need a solution how to keep the phone working at all time. regards, Digisoft VOIP

Make sure you use G.711u on all codec choice and do not use STUN or any nat traversal feature. What is the version of pbxnsip? Digisoft VOIP

Hello, One to many times in a day, the snom 360 and snom 320 will have a registration fail. Of course we have to reboot the phone. This is very anying because we never know when the fail will happen. PBX version 1.5.2.10a and snom FW 6.5.12 and 6.5.10 are used. we have many phones and we cannot get this to happen all the times. Any solution welcomed. Digisoft VOIP

Hello, Does anyone have suycessfully made a cisco 7961G work with pbxnsip 1.5.2.10a running on windows server 2000 We have a trace (pbxnsip): [0] 20071007123919: SIP Rx udp:200.123.47.50:18148: REGISTER sip:66.46.213.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.3.81:5080;branch=z9hG4bK3fcbb041 From: <sip:5149048083@66.46.213.1>;tag=0017e03419da000c2cc88fbb-e9a5f8ed To: <sip:5149048083@66.46.213.1> Call-ID: 0017e034-19da0002-73c8cff0-8dfe5212@192.168.3.81 Max-Forwards: 70 Date: Sun, 07 Oct 2007 16:39:19 GMT CSeq: 111 REGISTER User-Agent: Cisco-CP7961G/8.3.0 Contact: <sip:5149048083@192.168.3.81:5080;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0017e03419da>";+u.sip!model.ccm.cisco.com="30018" Supported: (null),X-cisco-xsi-6.0.1 Content-Length: 0 Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP0017E03419DA Load=SIP41.8-3-1S Last=Phone-Keypad" Expires: 3600 [0] 20071007123919: SIP Tx udp:200.123.47.50:18148: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.3.81:5080;branch=z9hG4bK3fcbb041;rport=18148;received=200.123.47.50 From: <sip:5149048083@66.46.213.1>;tag=0017e03419da000c2cc88fbb-e9a5f8ed To: <sip:5149048083@66.46.213.1>;tag=14990 Call-ID: 0017e034-19da0002-73c8cff0-8dfe5212@192.168.3.81 CSeq: 111 REGISTER Contact: <sip:5149048083@192.168.3.81:5080;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-0017e03419da>";+u.sip!model.ccm.cisco.com=30018;expires=90 Content-Length: 0 Even if we see in wireshark that the request is sent back to the phone, the phone never get the "X" removed and still show unregistered. Somehow pbxnsip thinks that the phone is registered. But will never be able to pass calls to or from the 7961G. The request sent to the phone seems to be on the wrong port number. (49157) the port is in fact 5080. we see that with wireshark 0.99.6. Digisoft VOIP