snom Canada
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Posts posted by snom Canada
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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:
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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:
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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
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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
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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.
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Ça ne fonctionne toujours pas. Voilà le résultat des tests:
--- PING 72.55.134.154 (72.55.134.154) 56(84) bytes of data. ---
--- 72.55.134.154 ping statistics ---
packets transmitted 9
received 0
packet loss 100 %
time 8063 ms
--- PING gw01.modulis.ca (72.55.134.154) 56(84) bytes of data. ---
--- gw01.modulis.ca ping statistics ---
packets transmitted 9
received 0
packet loss 100 %
time 8063 ms
traceroute to gw01.modulis.ca (72.55.134.154), 30 hops max, 60 byte packets
1 * * *
2 hos-tr4.juniper2.rz10.hetzner.de 213.239.224.97 de 4.250 ms
hos-tr2.juniper1.rz10.hetzner.de 213.239.224.33 de 0.139 ms 0.151 ms
3 hos-bb1.juniper4.ffm.hetzner.de 213.239.240.230 de 10.510 ms 10.514 ms
hos-bb1.juniper1.ffm.hetzner.de 213.239.240.224 de 4.810 ms
4 r1fra1.core.init7.net 77.109.135.17 ch 4.927 ms
r1fra1.core.init7.net 82.197.166.85 ch 12.969 ms 13.011 ms
5 r1fra2.core.init7.net 77.109.128.138 ch 5.027 ms 5.074 ms 5.165 ms
6 r1ams2.core.init7.net 77.109.128.201 ch 19.913 ms 19.847 ms 19.853 ms
7 xe-10-2-0.ams20.ip4.tinet.net 77.67.76.9 de 15.537 ms 15.550 ms 15.515 ms
8 xe-3-2-0.mtl10.ip4.tinet.net 89.149.181.25 de 105.013 ms 105.019 ms
xe-1-1-0.mtl10.ip4.tinet.net 89.149.184.74 de 102.976 ms
9 iweb-technologies-gw.ip4.tinet.net 173.241.130.106 us 105.306 ms 103.231 ms 105.370 ms
10 * * *
11 * * *
12 * * *
No reply for 3 hops. Assuming we reached firewall.
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.
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Rien à faire... mon routeur est bien configué, j'ai suivi la procédure de configuration du truck pour modulis et ça ne fonctionne toujours pas. J'ai encore et toujours le message d'erreur 408.
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?
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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
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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:
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White noise is usually caused by SRTP problems. Try turning it off on the phone and the PBX.
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)
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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,
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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
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Well, what you can do is run a PCAP trace on the phone and see if it is really sending REGISTER packets. Or if you can have Wireshark in the network you will get a "objective" view on where it is failing. If you filter by IP address the actual PCAP trace will not become too big, so that you can run it for several hours.
If you are running the phones behind NAT, it would also be very interesting if the packets arrive at the PBX unmodified. There is some equipment out there that tries to behave smart and patches SIP packets in a bad way or randomly changes ports. Just want to make sure we are not burning time on such bad equipment!
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
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If the phone does not register any more, I would recommend to set the transport layer to UDP and use IP addresses (no DNS addresses). If that is not stable, there is something really strange going on. If the DNS does the trick, check out the availability of the DNS server. If the UDP does make a difference, there msut be something with the stability of the TCP connections in this setup.
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,
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The problem here is that the Cisco phone sends UDP packets from a different port than it is listening on. This is extremly NAT unfriendly, and such a phone will never work behind NAT (FYI).
There is a setting on the phone called "NAT friendly UDP ports" or so that turns this off. If you toggle this flag the Cisco phone will use the port 5060 for sending and receiving.
Cisco is one of the few phones that do this. Practically all other phones use the same port for sending and receiving.
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.
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I would suggest to use the logging feature in the PBX. You can set per extension on what log level registration events should be logged. For example, if you set it to log level 1 and set the general log level to 2, you should be able to see when the phone looses it's registration and when it comes back.
This feature helped in other situations to figure out what was wrong. In that case there was a router that was changing the NAT binding from time to time and during that switch-over the registration got lost. A replacement of the router solved the problem.
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
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Thanks for your reply, Detlef.
I have checked most of that stuff out and to no avail. It seems to be a pbxnsip issue as we've had this issue before.
Not sure why it is but I'm going to also post in the main.
Ryan
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
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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
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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
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. 
ACD max callers problem
in Agent Group Setup
Posted
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)