secure sip trunk cucm driver
Hello
I have set up non-secure sip trunk using tcp of cucm to the driver
It's up on top of the cucm but in the driver call control is inaccessible
so the Conference made from an endpoint works but not the scheduled conference or composed auo
also, I set up a secure safe using tls
This time sip trunk is down in the side cucm and so it's on the driver side
I tried to download the temporary driver's certificate in the callmanager-trust and the tomcast-trust
Download the certificate of call manager to the conductor
but still does not work
I don't know if I need to generate a certificate authority certificates or just, I missed something?
CUCM: 10.5.2
conductor: XC4.0
Thank you
Hello
I recently deployed the CUCM and conductor (same version) as mentioned above.
My case was a little different as my certificate CUCM management was made by the internal certification authority, similarly to the conductor, I did the management of certificates, made sure the root certification authority is present in the two server to trust each other.
So I configured the location with port 5061 with ip address CM on the conductor.
Similarly on CM SIP trunk pointing to ad-hoc conductor and meeting woth port 5060 and security profile, device security mode an encrypted and the subject name X.509 to match the FQDN or Cluster COMPLETE domain name to match the domain the driver's FULL name, which allows the TLS communication code.
If I remember I was too faced the same problem when you face, however, after appropriate management of certificates and security profile, it has been resolved properly.
This guide should be useful
http://www.Cisco.com/c/dam/en/us/TD/docs/Telepresence/infrastructure/con...
Please let me know if you need more help.
Kind regards
RACLOT
Tags: Cisco Support
Similar Questions
-
PEI - SIP - CME - SIP - error CUCM Media is not Acceptable
Hello world
I have a problem with a TRUNK of SIP ITSP, the question is apparently "SIP/2.0 488 not acceptable media.
I tried several things, I Don t know how to solve this problem.
Outgoing calls is already OK, the problem is with incoming calls: of the ITSP to the CUCM.
I have this topology of the ITSP SIP TRUNK:
ITSP - sip sip - CME - CUCM
The CME configuration is:
Dial-peer voice voip 67
Description * SIP trunk ITSP *.
destination-model 591 [67]...
session protocol sipv2
session target ipv4:172.17.0.13
session udp transport
voice-class sip forced early offer
no interaction of dtmf
Codec g711ulaw!
Dial-peer voice voip 68
Description * SIP trunk CUCM *.
reply-to address. T
session protocol sipv2
session target ipv4:172.16.6.3
voice-class sip forced early offer
Codec g711ulaw!
SIP - ua
Disable-early-media 180
connection-reuse!
voip phone service
list of approved IP addresses
IPv4 0.0.0.0
IPv4 0.0.0.0 0.0.0.0
h323 connections allow h323
allow connections h323 to SIP
allow connections sip h323
allow sip to sip connections
no service additional h450.7
no additional service moved temporarily sip
no service additional sip refer
Fax protocol t38 ls-redundancy version 0 0 hs-redundancy 0 help none
SIP
binding control source-interface GigabitEthernet0/1.5
bind media source-interface GigabitEthernet0/1.5
Registrar Server
offer-early forced!
On the side of CUCM:
End point of media (checked)
Disable the media beginning on 180 (unchecked)
Requires the idle exchange of SDP for call Media Change (checked)
Early support for voice and video calls (checked)
Send send-receive SDP appealed INVITES (checked)The result of "debug messages ccsip" and "debug dialpeer inout voice" is:
001062: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
GUEST sip:[email protected]/ * /: 5060; user = phone SIP/2.0
Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
Call ID: [email protected]/ * /.
From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >
CSeq: 1 INVITE
Max-Forwards: 69
Contact:
Allow: INVITE, ACK, OPTIONS, CANCEL, INFO, BYE PRACK, NOTIFY, MESSAGE, UPDATE
P - asserted-Identity has:
Supported: 100rel, histinfo, prerequisite
P-early-Media: support
Content-Length: 362
Content-Type: application/sdpv = 0
o = HuaweiSoftx3000 1102026905 1102026906 IN IP4 172.17.0.11
s = SipCall
c = IN IP4 172.17.0.11
t = 0 0
m = audio RTP/AVP 8 18 116 10386
a = rtpmap:8 PCMA/8000
a G729/8000 rtpmap:18 =
a rtpmap:116 telephone-event/8000 =
a = ptime:5
a = sendrecv local curr:qos
a = distance zero curr:qos
a = sendrecv local compulsory are: qos
a = sendrecv distance optional with: qos
a = 3gOoBTC001063: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Number = 69200020, called number = 69200020, Peer Info Type = DIALPEER_INFO_SPEECH
001064: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
Associate the rule of = DP_MATCH_DEST; Called number = 69200020
001065: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
No outbound dial-peer does not; Result = NO_MATCH(-1)
001066: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring = 69200020, saf_enabled = 1 saf_dndb_lookup = 1, dp_result =-1
001067: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
Result = NO_MATCH(-1)
001068: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Number = 70965999, called number =, Voice-Interface = 0 x 0.
Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
Peer Type Info = DIALPEER_INFO_SPEECH
001069: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
001070: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
001071: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Number = 70965999, called number =, Voice-Interface = 0 x 0.
Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
Peer Type Info = DIALPEER_INFO_SPEECH
001072: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
001073: 03:31:03: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
001074: 03:31:03: //-1/CD12136099D5/DPM/dpAssociateIncomingPeerCore:
Number = 70965999, called number = 69200020, Voice-Interface = 0 x 0.
Timeout = TRUE, Peer Encap Type = ENCAP_VOIP, Type of research peer = PEER_TYPE_VOICE,
Peer Type Info = DIALPEER_INFO_SPEECH
001075: 03:31:03: //-1/CD12136099D5/DPM/dpAssociateIncomingPeerCore:
Result = Success (0) after DP_MATCH_ANSWER; Incoming dial-peer = 68
001076: 03:31:03: //-1/CD12136099D5/DPM/dpMatchSafModulePlugin:
dialstring = NULL, saf_enabled = 0, saf_dndb_lookup = 0, dp_result = 0
SIP: Attempt to analyze the attribute not supported at the level of the media
001077: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Envoy:
SIP/2.0 488 Media is not Acceptable
Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >; tag = C139A0-1755
Date: Thu, November 6, 2014 13:01:04 GMT
Call ID: [email protected]/ * /.
CSeq: 1 INVITE
Allow-events: telephone-event
Reason: Q.850; cause = 65
Server: Cisco-SIPGateway/IOS-15.3.3.M2
Content-Length: 0001078: 03:31:03: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:SIP GatewayTelf_JD #ACK:[email protected]/ * /: 5060; user = phone SIP/2.0
Via: SIP/2.0/UDP 172.17.0.11:5060; branch = z9hG4bKi2mhl410do70mrc4s141.1
CSeq: 1 ACK
Call ID: [email protected]/ * /.
From: "70965999"<>[email protected]/ * /; transport = udp; user = phone >; tag = 2u2uavrx-CC-1013
To: '69200020'<>[email protected]/ * /; transport = udp; user = phone >; tag = C139A0-1755
Max-Forwards: 69
Content-Length: 0Little light at the end of the tunnel?
Thanks in advance!
Hello
can you collect debug voice ccapi inout & debugging ccsip GCE message and attach the logs here please?
your provider sends A Law G711 codec in the PROMPT message, but you have configured G711 U right in the dial-peers.
can you try to fix G711 has the right dial-peers and check out them? Also make sure you have TPMS in the MRGL applied to CUCM SIP Trunk.
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I CUCM connected to three different Lync server via 3 different SIP trunks.
RG is composed of the following elements:
LYNC SIP TRUNK 1 (1.1.1.1)
LYNC SIP TRUNK 2 (2.2.2.2)
LYNC SIP TRUNK 3 (3.3.3.3)
The route group was built with "Top Down" the algorithm of distribution. The first SIP trunk knows congestion and some calls are never routed to secondary and tertiary SIP trunks.
Based on all the forum posts I've seen - it seems that I have to configure the algorithm of group distribution of ranges as 'circular '.
If I change the algorithm group of "Circular" lines - can I expect the following results:
1. first call will go through LYNC SIP TRUNK 1
2. second call will cross LYNC SIP TRUNK 2
3. third call will cross LYNC SIP TRUNK 3
When I change the algorithm of distribution of the route to the 'circular' group and click 'SAVE', I am invited on "RESET". This service assigns to the existing calls through SIP Trunks?
TIA,
Amir
Hello
the circular algorithm will work the way you mentioned, but I suggest to try and make sure that is not perform integration
to reset the SIP trunk actually active calls should not be made because the gose media directly between endpoint unles syou use something like trust rely wher evous enforce calls to go to the SIP
Therefore, in most cases is should be fine just try configuration at the time of the calls will face service disruption
hope this helps
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Problem with a SIP Trunk between VCS and CUCM
Hello world
I created a SIP Trunk between control VCS (X7.1) and a CUCM 8.6.
I followed this Cisco_VCS_Cisco_Unified_Communications_Manager_Deployment_Guide_CUCM_6-1_7_8_and_X7 - 0.pdf deployment guide.
The trunk is in place, but I have the following problem: I can ring since an EX90 recorded on the VCS for a registered on the CUCM 9951 but when I take the call, I have a busy signals (audio and video). When I on the EX90 9951 ring, I have the following message on VC: can not reach the contact.
I have a 503 service unavailable status on the call history and "call rejected" on the event log.
My transformation is OK and the call is sent to my neighbor CUCM zone
You have an idea to solve this problem?
Thanks for your help
Best regards
Bruno Z.
Hello
Yes, it's true! It seems at first glance a problem with codec negotiation when I see the reason 47 code.
in any case set the region to G.711 trunk phone in case if you have different device pool and the trunk region and ip-phones.
make G.711 end to end and test again.
Thank you
Alok
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is it possible, end point of video recorded in CUCM call standalon SX10 via SIP Trunk?
Dear all,
Need help :)
1.
is it possible, end point of video recorded in CUCM call standalon SX10 via SIP Trunk?
my customers will not buy TP - lic :)
Topology:
8900 (jabber) - CUCM-(SIP) - SX10
2.
Customer, have MCU machine.
If not possible. is it possible, meeting point of MCU use?
Topology:
8900 (jabber) - CUCM - MCU - SX10
1. Yes, but it is not that simple and straightforward that the SX10 can be reached by calling is the public IP address and CUCM does not support the IP address numbering is not a SIP address format...
2. Yes, using the MCU would be much simpler, easier and, in my opinion, a better solution to be implemented. Here is an example of how a stand-alone SX10 could dial the IP address of the MCU standard automatic.
You can also assign a SIP URI address to the meeting point which the SX10 can connect to.
/Jens
Please note the answers and score the questions as "answered" as appropriate.
-
I would like to know if it is possible to create a Script of SIP standardization in CUCM which change the Destination SIP port based on the phone which makes the phone.
Current issue:
The ITSP provider asked that I send calls from different regions to the same address IP SBC but a different port.
Example:
Phone (A) makes a call
RP - RL - RG---> SIP Trunk (5060)---> CUBE (5060<-->5001)---> ITSP1
Telephone (B) makes a call
RP - RL - RG---> SIP Trunk (5060)--->---> CUBE ITSP1 (5060<-->5002)
I can do it on the CUBE, but which require several dial-Exchange and SIP profiles, is it possible in the CUCM to change the sip by device-pool port?.
Or is there another I can match and send calls via the same destination with different dial-peer sip ports.?
Thank you.
Zakiab,
You can do this, its all just impossible. Your signage ports are defined not on the endpoints, but on the B2BUA (CUCM or CUBE). The CUBE, you have the most flexibility, because you can change the port of signs based on your dial-peers. However, as you pointed out quite rightly, need you an insurmountable amount of dial-peers to have this for each endpoint. So I suggest tell you your ITSP to change their design or moving to a new.
-->--> -
Hello
I have a requirement to pass an h323 to SIP environment environment. I'm looking for good practices, especially around security. I have 2 servers CUCM (8.5) in cities separated for redundancy. I have also 2 voice gateways which, at the present time, h323 to the PSTN, are each located in different cities.
My requirements are:
1. create a sip trunk instead of the supplier of the use of PRI.
2 If the Wan link fails on a gateway provider, router replacing in the other location should be able to receive installation messages and if a user connects via extension mobility, should be able to answer the call.
Is there a simplified design docos on for this? I hesitate to create a SIP trunk directly to the supplier for safety, thus thinking to end the call on the routers of voice with the CUBE. I am sure that it is managed from the factory and would appreciate comments.
See you soon!
Pieter
Simple answer use ALWAYS the CUBE. With IOS 15.1 T and more you have security against fraud free of charge that you can use to restrict which can address IP contacted the CUBE, that's all you need.
HTH,
Chris
-
CallManager Express SIP trunk problem
Hi all here.
I have a problem with my SIP trunk CallManager Express (version 10.0). On my site already configured trunk SIP between CME and CUE. I have configured DN for SIP phone and SIP phones recorded on CME successfully, but... As soon as SIP phone registered on CME, DIAL-PEER for CME and CUE connection changes. Especially 'session target ipv4' automatically changes to my IP phone SIP and all calls go to my SIP phone and does not reach the auto attendant.
How can I solve this problem?
Hello
Following is the error of CUCM;
WARNING: 399 UnicompCM "cannot find a Device Manager for the request received on port 55549 leave 192.168.10.2.
This probably means that CME use 192.168.10.2 for source packages SIP CUCM however SIP trunk in CUCM is not directed to this IP address, it is useful to point out to another IP interface of the GUY and therefore dismiss this appeal.
Can you please check what is the IP of CMF address you configured on the SIP trunk in CUCM?
-Vivek
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Configure (fxs) analog phones with Sip trunk
Hi all
IS - this configuration FXS possiple to knit on SIP TRUNK? I HAVE 16 FXS ports and voip gateway 2921 cisco.
Configuring fxs to work with her?
any help thanks.
Is there a PBX at stake here, i.e. CUCM or CME?
In both cases, you have set good dial-peers to point to the FXS ports to match the destination of the SIP trunk, which is pretty simple.
-
Hello
I have a problem in case of detection of the DTMF
We have a SIP of the ITSP Trunk and everything is ok except DTMF.
The sip trunk is between ITSP and router 3945
ITSP <->3945 <->CUCM 10.5
I have try all method such as rtp - nte, h245 alphanumeric and h245-signal, info-sip, sip-kpml,... in line-peer to ITSPs
ITSP say that he send with standard RFC2833 dtmf (it equals to rtp-net, I know), but when I am 'debug ccsip message', the results show the dtmf with rfc2833 sends us
16 August 13:52:28.991: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:Received:INVITE sip: [email protected] / * /: 5060. user = phone SIP/2.0Via: SIP/2.0/UDP 10.105.40.34:5060; branch = z9hG4bKejhh8jixkobhnb7ykvi87vuj8Call ID: [email protected]/ * /.From: sip: [email protected] / * />; tag = c3cx3vcw-CC-40To: sip: [email protected] / * /; user = phone >CSeq: 1 INVITEAllow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTRY, PRACK, INFO SUBSCRIBE, NOTIFY, updates, MESSAGE, seeMax-Forwards: 69Supported: 100rel, timerUser-Agent: Huawei SoftX3000 V300R010Session time-out: 300Min - SE: 90Contact: sip: [email protected] / * /: 5060; user = phone >Content-Length: 374Content-Type: application/sdpv = 0o = HuaweiSoftX3000 11042757 11042757 IN IP4 10.105.40.34s = call Sipc = IN IP4 10.105.40.34t = 0 0m = audio 27762 RTP / AVP 8 0 18 4 2 98 99 102a = rtpmap:8 PCMA/8000a = rtpmap:0 PCMU/8000a G729/8000 rtpmap:18 =a = rtpmap:4 G723/8000a = rtpmap:2 G726-32/8000a = rtpmap:98 G726-40/8000a = rtpmap:99 G726-32/8000a = rtpmap:102 G726-24/8000a = ptime:20a = fmtp:18 annex b = No.It is a message to guest (with sdp) of ITSPAs you can see the line with red color must have a code with number of 101 but rather a code with number of 18In my "ccsip media debug' output show that the method of negotiation between me and itsp 'incoming voice. 'It's my router config:voip phone service
No IP trust to authenticate
allow connections h323 to SIP
allow connections sip h323
allow sip to sip connections
SIP
interface FastEthernet0/0/1 source control binding
bind media source interface FastEthernet0/0/1
min - to 300 session expires-300
!Dial-peer voice 2 voip---> router CUCM and vice versa
translation-profile outgoing toos
destination-model 42584...
session protocol sipv2
session target ipv4:10.20.30.70
Codec g711ulaw
DTMF-relay rtp - nte
!
VoIP voice 10 Dial - peer---> router for ITSP and vice versa
destination-model. T
session protocol sipv2
session target ipv4:10.105.40.34
incoming called-number. T
DTMF-relay rtp - nte
Codec g711ulawI have configured cucm with a sip section to my favorite router with active PSG and RFC2833BUT HE DIDN'T THERE WAS NO DETECTION OF DTMF IN INCOMING CALLS AND OUTGOINGI even test dial-position 10 without any method of dtmf-relay because I want to configure it with the incoming voice method, but it does not workI change the codec but does not solve the problemThere is an interesting point and that is, if use Elastix 3945 of the router and configure dtmf among elastix and itsp as "inbound" method and method dtmf among elastix and cucm as 'rfc2833' everything is OK (ITSP<--Inbound--> Elastix <--rfc2833-->CUCM)--rfc2833--> --Inbound-->Please give me a solution to solve the problem between Cisco 3945 and ITSPConcerning->->It would be more accurate to say that the problem has been bypassed by using a transcoding resource. The correct resolution here would be to open a ticket to trouble with the ITSP. I agree that the PROMPT message you show doesn't include advertising RFC2833 in the SDP offer. It is a problem that only the carrier can rectify.
-
SIP trunk CUBE with Callcentric - incoming unanswered call
I'm doing some tests with a Sip trunk with a provider called Callcentric.It is a CUBE scenario. I use a SIP to the CUCM trunk.
I have a Cisco Callmanager and a virtual router 7200 (GNS3) as a gateway (C7200-ADVENTERPRISEK9-M), Version 12.4 (24).A CPIC connected to Callmanager, I call out to PSTN and it works perfectly.
When I do an incoming call to the PSTN to the softphone destination, she sounds, but when I press the answer button, the call is not connected and the analog phone in RTC can listen back ringtone.
Do you have any idea what it could be?
Some relevant configurations:voip phone service
allow sip to sip connections
Fax protocol cisco
SIP
interface FastEthernet0/0 source control binding
bind media source interface FastEthernet0/0
Registrar Servervoice class codec 1
g711ulaw codec preference 1translation of the voice-rule 1
rule 1 / ^ 8 / /0056/
!
voice translation-rule 2
rule 1 5.0 / /17772114zzz/
!
voice translation-rule 3
rule 1 /17772114zzz/ /500/voice translation-profile IN
definition of 3 called
!
FLIGHT voice translation-profile
definition of call 2
translate 1 calledDial-peer voice 1 voip
CALLCENTRIC description
entrants IN translation-profile
translation-profile outgoing OUT
destination-model 8.T
codec voice-class 1
session protocol sipv2
session target sip-Server
incoming called-number 17772114zzz
SIP DTMF-relay-notify rtp - nte
!
Dial-peer voice 2 voip
CUCM description
destination-model 500
media stream-autour
codec voice-class 1
session protocol sipv2
session target ipv4:192.168.10.116
incoming called number 8.T
SIP DTMF-relay-notify rtp - nteSIP - ua
credentials of username, password 7 17772114zzz 094F471B100928 callcentric.com Kingdom
authentication username 17772114zzz password 7 121A0C051B0803 callcentric.com Kingdom
no remote-party-id
Registrar dns:callcentric.com expires 3600
DNS:callcentric.com SIP server
Home-Office
Thank you guys.Hello
Please configure the sip requires low profile to remove the header and apply this profile in the dial-peer 1 outgoing.
SIP-class voice profiles 1
response header 200 sip requires DELETE
If this does not work under Dial-peers, try to apply globally.
voip phone service
SIP
SIP profiles 1
Suresh
Please note all useful posts
-
Anyone know if this is in the roadmap an improvement on SIP/security settings in the ISDN GW 3241?
We where to check the config and we couldn t find a way to configure the GW (version 2.2 (1.79) P to use SIP and avoid using non desired.)
When the Dial Plan (IP ISDN) is configured as SIP or other, there is no configuration in the Don t Gulf war do not accept calls from any device that sends an INVITATION. It takes a SIP trunk configuration, to make a relationship with her pair (only accept call form SIP trunk sources, certificates, etc.).
Any idea?
It is an impact on deployment of client (Government).
Thank you
Ok. Thanks for the answer (of course side would be also appreciated :-) of marking
Not to mention that sometimes some information of the road map are mentioned here I would not wait or wait until it
as this is a public forum, but the road map info is often under NDA.
I recommend you talk to your partner about Cisco / contact and see if you can get a presentation of the road map
and also note the impact for your deployments and possible feature requests for them.
Hope that answered your question :-)
Martin
-
CME: assign extension sip trunk
Hello
Instead of using a prefix to use a separate sip trunk, I would like an IP phone to use a separate sip for its 2nd line trunk.
Then I set up a 2nd line on an IP phone to use extention 96:
ePhone 1
Mac address *.
name *.
button 2:96ePhone-dn 96 double line
Number of 432and then I would this extension (with the number ending in 432) always use a SIP server.
I think I need a dial-peer to achieve this, similar to:
Dial-peer voice voip 432
destination-model?
codec voice-class 1
voice-class sip dtmf-relay rtp - nte force
session protocol sipv2
session target sip-Server
DTMF-relay rtp - nte
No vadHow can I join the dial-peer name extension? (for analog, I would put "monitor trunk 1 * voice port number *"on the name extension ").
Any help appreciated,
Jonathan
Hello.
You can try with answer address under your dial-position 432 432.
HTH
Concerning
Carlo
-
SIP trunk behind a router using NAT
Hello
Is it possible to use a SIP trunk to a provider SIP ITSP having the CUBE / router gateway behind a firewall using a NAT?
Does anyone do this?
I ask because I'm having problems to make my SIP trunk to work and my router for cube is behind my generic service provider router, which makes the NAT. I just want to rule this out as a problem.
Has anyone else done this? Or is it really impossible?
Thank you very much
Tom
Hello
As NAT works fine SIP would work properly as the Protocol.
Here is the RFC for "NAT Traversal practices for Client - Server SIP"
https://Tools.ietf.org/html/rfc6314
HTH
JB
-
Source IP address of the originating call to sip trunk
I'm unable to set address ip source from trunk sip call. I read somewhere that the source ip address of the call must be the ip address that call device registered to. I also checked "run in all active nodes such as suggested by cisco. We have 3 Sub cluster and 1 pub however still call orgianate to a particular Sub.
I also undergoes several changes i - e DP and list route but think not worked for me. Can someone help to sortout this problem?
Let's say if your ip phone registered in sub - and "run on all nodes" enabled, then the source ip address is always sup - A s ip address regardless of the device pool config on the sip trunk.
Could please explain you the device pool config in phones IP & SIP trunk?
you have enabled the "run on all nodes" for a list of course also?
Suresh
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After authentication check after user authentication using authentication SSO OAM
Hi allWe have recently configured all our apex oracle with OAM SSO application. Authentication works fine but the problem is, after the connection of users, we redirect users to different pages of the application based on their user role that is defi