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- About corneliss
07-26-2019
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Member since
08-11-2014
87
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11
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2
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Created by
corneliss
in IP Telephony and Phones
in IP Telephony and Phones
07-24-2019
09:35 AM
07-24-2019
09:35 AM
Hi Jinto, I will try to get this to you as RTMT takes forever to load the messages and I need to remove IP's Can you maybe specify which ones are needed most so I can get it to you, or is all the messages needed? As Kevin also mentioned, we are definitely using t.38 on the FAX Server as I did see RTP messages on a call that worked and a few t.38 messages coming through. No t.38 messages are present when a call comes from the SIP Cube.
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Created by
corneliss
in IP Telephony and Phones
in IP Telephony and Phones
07-24-2019
09:12 AM
07-24-2019
09:12 AM
Hi Kevin, Yes it does use T.38 I did see the T.38 request coming through when doing a test from a PRI where Fax is working. I did also then notice that G711alaw PCM was used in the Wireshark traces and that's why I was under the impression that I might be sending the wrong Codec. When adding no modem passthrough to my Dial Peer from Cube to CUCM the fax still fails.
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Created by
corneliss
in IP Telephony and Phones
in IP Telephony and Phones
07-23-2019
10:03 PM
07-23-2019
10:03 PM
Hi Jinto Sorry about that, let me explain quick. First Line is my CUBE Router 192.168.1.10 Second line is my First Subscriber 192.168.1.11 3rd Line is my Second Subscriber 192.168.1.12 - Not sure why it's going to both Subscribers 4th Line is the connection to the RightFax Server over a SIP Trunk - lets say 192.168.1.50 is the IP. Please note we did have two Right Fax Servers for Redundancy and one is currently offline, there is also two SIP Trunks Pointing to Both RightFax Server. When Testing a working solution to Right Fax as seen below the call only hits the one Subscriber and Routes successfully to my Fax Server. Subscriber 1 --------------- RightFax Server This is when dialing the Fax number from a cellphone.
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Created by
corneliss
in IP Telephony and Phones
in IP Telephony and Phones
07-23-2019
07:53 AM
07-23-2019
07:53 AM
Hi Jinto, Please see attached is the error from RTMT.
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Created by
corneliss
in IP Telephony and Phones
in IP Telephony and Phones
07-23-2019
07:10 AM
07-23-2019
07:10 AM
Hi Jinto, I have tried the following you have suggested as before but the issue always remains. Running a Pcap on the Fax Server I get the following 488 not acceptable media -does not matter if it's g711 or g729
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Created by
corneliss
in IP Telephony and Phones
in IP Telephony and Phones
07-22-2019
11:35 PM
07-22-2019
11:35 PM
Hi, So under Global Config: voice service voip ip address trusted list mode border-element allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none sip Dial Peer incoming Fax number from ISP ----- CUBE dial-peer voice 101 voip description ##Inbound Calls from SIP## translation-profile incoming IS-SIP-PWC-CUBE session protocol sipv2 incoming called-number 27127XXX... (hidden) voice-class codec 2 (Changed this to a class that support g711ulaw and alwaw to see if this fixes the issue) Generally it sets to g729 dtmf-relay rtp-nte sip-kpml no vad Dial Peer incoming Fax number from CUBE-----CUCM dial-peer voice 12 voip description ##Incoming_CUCM## preference 1 destination-pattern 27127XXX... (hidden) session protocol sipv2 session target ipv4:10.XX.XX.XX voice-class codec 2 (Changed this to a class that support g711ulaw and alwaw to see if this fixes the issue) Generally it sets to g729 voice-class sip bind control source-interface Loopback0 voice-class sip bind media source-interface Loopback0 dtmf-relay rtp-nte sip-kpml
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Created by
corneliss
in IP Telephony and Phones
in IP Telephony and Phones
07-20-2019
02:35 AM
07-20-2019
02:35 AM
Hi All, I need some help, we have recently started moving from MGCP Gateways to SIP CUBE on our Routers as the PRI quality was not good. I have Ported my Fax Range to the new ISP and currently having trouble receiving Faxes when it hits my CUBE. Please we have a SIP Trunk between CUCM and our RightFax Server and I can confirm is working for all MGCP Gateways. When doing a call to my FAX number using the CUBE I do hear the Fax tones but the call is dropped after 3 seconds. This means all routing is correct but it seems like connection is dropping. I have read online it might be because I'm sending g729 and g711ulaw to my Fax but can confirm when changing the Codec the call fails instantly. I have not worked with Faxing before and it's extremely hard to troubleshoot. any suggestions or ways to use RTMT to see what the issue is. Or has someone else using RightFax used their CUBE Gateway for Faxing and got it working. Thanks in advance.
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Labels:
07-27-2017
04:26 AM
Hi Jitender,
Thanks for this, I would also like to confirm.
I see in the article this was more for people that used the Router as an MCU.
All we need to do is ISDN Bondings. We are trying to move away from the MSE 8000 and the ISDN 8321 for our ISDN Audio and Video Calls.
Do you know of any alternative way to do ISDN calls to Video?
Or will we need to buy a new ISDN Gateway.
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Created by
corneliss
in Other Collaboration Subjects
in Other Collaboration Subjects
07-26-2017
05:55 AM
07-26-2017
05:55 AM
Hi,
I have recently set up a H323 Gateway (using a Cisco Router) in my environment for the purpose of using it as the gateway for my VCS.
Audio is working just fine but I need to set it up to allow for video calls over ISDN.
I am setting it up on a 3945 ISR router with PVDM3-256 installed.
What all do I need to configure to get this to work? There is not much information I could find online.
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Labels:
07-13-2017
11:03 AM
Hi Manish,
We have sorted out incomming Voice Calls on the Router. We are now struggling with the Video portion, to make Video Calls over ISDN. Do you have any idea on how to do this or something similar.
Jul 13 18:01:10.775: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0D1B Sending Complete Bearer Capability i = 0x8890 Standard = CCITT Transfer Capability = Unrestricted Digital Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA18394 Preferred, Channel 20 Progress Ind i = 0x8083 - Origination address is non-ISDN Calling Party Number i = 0x2183, '0112189300' Plan:ISDN, Type:National Called Party Number i = 0xC1, '0112029999' Plan:ISDN, Type:Subscriber(local) Jul 13 18:01:10.775: ISDN Se0/0/0:15 Q931: Received SETUP callref = 0x8D1B callID = 0x00E9 switch = primary-net5 interface = User ZATJHBCUGPUX003# Jul 13 18:01:10.779: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8D1B Channel ID i = 0xA98394 Exclusive, Channel 20 Jul 13 18:01:10.939: ISDN Se0/0/0:15 Q931: TX -> ALERTING pd = 8 callref = 0x8D1B ZATJHBCUGPUX003# Jul 13 18:01:20.979: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x8D1B Cause i = 0x80AF - Resource unavailable, unspecified Jul 13 18:01:21.147: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x0D1B Jul 13 18:01:21.147: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x8D1B
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07-13-2017
04:39 AM
Hi Manish,
We use 5 digit DN's in our CUCM environment. The Gateway significant digits is set to 5. 29999 is a DN that exists in CUCM.
29999 is in a "none" partition. The gateway is able to reach it.
Our codecs do match and I have Transcoding and MTP dspfarm profiles on the router anyway that are registered and applied to the GW.
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Created by
corneliss
in IP Telephony and Phones
in IP Telephony and Phones
07-12-2017
11:50 PM
07-12-2017
11:50 PM
We are having issues setting up an H323 Gateway on our 3945 Router. To route calls from my PSTN to CUCM.
I have found multiple documents, relating to this issue, we have also updated it to be exactly the same.
The ip address on CUCM and my gateway loop back match and port 1720 is open.
I want to set it up to only accept calls not to dial out on hence only one dial peer.
Thanx
Running config:
!
voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback none h323 no h225 timeout keepalive ! voice class h323 1 h225 timeout tcp establish 3
!
interface Loopback0 ip address 10.20.44.7 255.255.255.255 h323-gateway voip interface h323-gateway voip bind srcaddr 10.20.44.7
!
interface Serial0/0/0:15 no ip address encapsulation hdlc isdn switch-type primary-net5 isdn incoming-voice voice no cdp enable
!
dial-peer voice 100 voip preference 1 destination-pattern .T session target ipv4:10.25.1.50 dtmf-relay h245-alphanumeric codec g711ulaw no vad
====================================================================================================== debug ip tcp transaction ====================================================================================================== Jul 12 19:35:56.748: TCB10324200 created Jul 12 19:35:56.748: TCB10324200 setting property TCP_NO_DELAY (0) 2EE55AC Jul 12 19:35:56.748: TCB10324200 setting property TCP_TOS (11) 2EE55D0 Jul 12 19:35:56.748: TCB10324200 setting property TCP_NONBLOCKING_WRITE (10) 2EE55E8 Jul 12 19:35:56.748: TCB10324200 setting property TCP_NONBLOCKING_READ (14) 2EE55E8 Jul 12 19:35:56.748: TCB10324200 setting property TCP_KEEPALIVE (17) 2EE55E8 Jul 12 19:35:56.748: TCP: Setting Keepalive interval and retries to 60 and 4 Jul 12 19:35:56.748: TCB10324200 setting property TCP_VRFTABLEID (20) 1391BC68 Jul 12 19:35:56.748: TCB10324200 setting property TCP_NO_DELAY (0) 2EE55D8 Jul 12 19:35:56.748: tcp_uniqueport: using ephemeral max 65535 Jul 12 19:35:56.748: TCP: Random local port generated 18087, network 1 Jul 12 19:35:56.748: TCB10324200 bound to 10.20.44.7.18087 Jul 12 19:35:56.748: Reserved port 18087 in Transport Port Agent for TCP IP type 1 Jul 12 19:35:56.748: TCB10324200 getting property TCP_STRICT_ADDR_BIND (19) Jul 12 19:35:56.748: TCP: sending SYN, seq 1489956361, ack 0 Jul 12 19:35:56.748: TCP0: Connection to 10.25.1.50:1720, advertising MSS 536 Jul 12 19:35:56.748: TCP0: state was CLOSED -> SYNSENT [18087 -> 10.25.1.50(1720)] Jul 12 19:35:58.748: TCP0: RETRANS timeout timer expired Jul 12 19:35:58.748: 10.20.44.7:18087 <---> 10.25.1.50:1720 congestion window changes Jul 12 19:35:58.748: cwnd from 536 to 536, ssthresh from 65535 to 1072 Jul 12 19:35:58.748: TCP0: timeout #1 - timeout is 4000 ms, seq 1489956361 Jul 12 19:35:58.748: TCP: (18087) -> 10.25.1.50(1720) Jul 12 19:36:02.748: TCP0: RETRANS timeout timer expired Jul 12 19:36:02.748: TCP0: timeout #2 - timeout is 8000 ms, seq 1489956361 Jul 12 19:36:02.748: TCP: (18087) -> 10.25.1.50(1720) Jul 12 19:36:10.748: TCP0: RETRANS timeout timer expired Jul 12 19:36:10.748: TCP0: timeout #3 - timeout is 16000 ms, seq 1489956361 Jul 12 19:36:10.748: TCP: (18087) -> 10.25.1.50(1720) Jul 12 19:36:11.748: Released port 18087 in Transport Port Agent for TCP IP type 1 delay 240000 Jul 12 19:36:11.748: TCP0: state was SYNSENT -> CLOSED [18087 -> 10.25.1.50(1720)] Jul 12 19:36:11.748: TCB 0x10324200 destroyed
======================================================================================================
debug isdn q931 ====================================================================================================== Jul 12 19:37:48.214: ISDN Se0/0/0:15 Q931: RX <- SETUP pd = 8 callref = 0x0EA1 Sending Complete Bearer Capability i = 0x9090A3 Standard = CCITT Transfer Capability = 3.1kHz Audio Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA1839A Preferred, Channel 26 Progress Ind i = 0x8281 - Call not end-to-end ISDN, may have in-band info Calling Party Number i = 0x2183, '0723745279' Plan:ISDN, Type:National Called Party Number i = 0xC1, '0112029999' Plan:ISDN, Type:Subscriber(local) Jul 12 19:37:48.214: ISDN Se0/0/0:15 Q931: Received SETUP callref = 0x8EA1 callID = 0x0053 switch = primary-net5 interface = User Jul 12 19:37:48.218: ISDN Se0/0/0:15 Q931: TX -> CALL_PROC pd = 8 callref = 0x8EA1 Channel ID i = 0xA9839A Exclusive, Channel 26 Jul 12 19:38:03.218: ISDN Se0/0/0:15 Q931: TX -> DISCONNECT pd = 8 callref = 0x8EA1 Cause i = 0x80A6 - Network out of order Jul 12 19:38:03.378: ISDN Se0/0/0:15 Q931: RX <- RELEASE pd = 8 callref = 0x0EA1 Jul 12 19:38:03.378: ISDN Se0/0/0:15 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x8EA1
======================================================================================================
====================================================================================================== debug voip ccapi inout ====================================================================================================== Jul 12 19:44:10.279: //-1/4F1E0A638058/CCAPI/cc_api_display_ie_subfields: cc_api_call_setup_ind_common: cisco-username= ----- ccCallInfo IE subfields ----- cisco-ani=0723745279 cisco-anitype=2 cisco-aniplan=1 cisco-anipi=0 cisco-anisi=3 dest=0112029999 cisco-desttype=4 cisco-destplan=1 cisco-rdie=FFFFFFFF cisco-rdn= cisco-rdntype=-1 cisco-rdnplan=-1 cisco-rdnpi=-1 cisco-rdnsi=-1 cisco-redirectreason=-1 fwd_final_type =0 final_redirectNumber = hunt_group_timeout =0
Jul 12 19:44:10.279: //-1/4F1E0A638058/CCAPI/cc_api_call_setup_ind_common: Interface=0x161B8070, Call Info( Calling Number=0723745279,(Calling Name=)(TON=National, NPI=ISDN, Screening=Network, Presentation=Allowed), Called Number=0112029999(TON=Subscriber, NPI=ISDN), Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Incoming Dial-peer=0, Progress Indication=NOT END TO END ISDN(1), Calling IE Present=TRUE, Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=-1 Jul 12 19:44:10.279: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 12 19:44:10.279: :cc_get_feature_vsa malloc success Jul 12 19:44:10.279: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 12 19:44:10.279: cc_get_feature_vsa count is 1 Jul 12 19:44:10.279: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 12 19:44:10.279: :FEATURE_VSA attributes are: feature_name:0,feature_time:271239848,feature_id:183 Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/cc_api_call_setup_ind_common: Set Up Event Sent; Call Info(Calling Number=0723745279(TON=National, NPI=ISDN, Screening=Network, Presentation=Allowed), Called Number=0112029999(TON=Subscriber, NPI=ISDN)) Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/cc_process_call_setup_ind: Event=0x2EB9678 Jul 12 19:44:10.279: //-1/xxxxxxxxxxxx/CCAPI/cc_setupind_match_search: Try with the demoted called number 0112029999 Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccCallSetContext: Context=0x31BDCB8 Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/cc_process_call_setup_ind: >>>>CCAPI handed cid 183 with tag 0 to app "_ManagedAppProcess_Default" Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccCallProceeding: Progress Indication=NULL(0) Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccCallSetupRequest: Destination=, Calling IE Present=TRUE, Mode=0, Outgoing Dial-peer=100, Params=0x31C4978, Progress Indication=NOT END TO END ISDN(1) Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccCheckClipClir: In: Calling Number=0723745279(TON=National, NPI=ISDN, Screening=Network, Presentation=Allowed) Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccCheckClipClir: Out: Calling Number=0723745279(TON=National, NPI=ISDN, Screening=Network, Presentation=Allowed) Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccCallSetupRequest: Destination Pattern=.T, Called Number=0112029999, Digit Strip=FALSE Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccCallSetupRequest: Calling Number=0723745279(TON=National, NPI=ISDN, Screening=Network, Presentation=Allowed), Called Number=0112029999(TON=Subscriber, NPI=ISDN), Redirect Number=, Display Info= Account Number=, Final Destination Flag=TRUE, Guid=4F1E0A63-6671-11E7-8058-7426AC319D80, Outgoing Dial-peer=100 Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/cc_api_display_ie_subfields: ccCallSetupRequest: cisco-username= ----- ccCallInfo IE subfields ----- cisco-ani=0723745279 cisco-anitype=2 cisco-aniplan=1 cisco-anipi=0 cisco-anisi=3 dest=0112029999 cisco-desttype=4 cisco-destplan=1 cisco-rdie=FFFFFFFF cisco-rdn= cisco-rdntype=-1 cisco-rdnplan=-1 cisco-rdnpi=-1 cisco-rdnsi=-1 cisco-redirectreason=-1 fwd_final_type =0 final_redirectNumber = hunt_group_timeout =0
Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccIFCallSetupRequestPrivate: Interface=0x272870C, Interface Type=1, Destination=, Mode=0x0, Call Params(Calling Number=0723745279,(Calling Name=)(TON=National, NPI=ISDN, Screening=Network, Presentation=Allowed), Called Number=0112029999(TON=Subscriber, NPI=ISDN), Calling Translated=FALSE, Subscriber Type Str=RegularLine, FinalDestinationFlag=TRUE, Outgoing Dial-peer=100, Call Count On=FALSE, Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=) Jul 12 19:44:10.279: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 12 19:44:10.279: :cc_get_feature_vsa malloc success Jul 12 19:44:10.279: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 12 19:44:10.279: cc_get_feature_vsa count is 2 Jul 12 19:44:10.279: //-1/xxxxxxxxxxxx/CCAPI/cc_get_feature_vsa:
Jul 12 19:44:10.279: :FEATURE_VSA attributes are: feature_name:0,feature_time:271237384,feature_id:184 Jul 12 19:44:10.279: //184/4F1E0A638058/CCAPI/ccIFCallSetupRequestPrivate: SPI Call Setup Request Is Success; Interface Type=1, FlowMode=1 Jul 12 19:44:10.279: //184/4F1E0A638058/CCAPI/ccCallSetContext: Context=0x31C4928 Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccSaveDialpeerTag: Outgoing Dial-peer=100 Jul 12 19:44:10.279: //184/4F1E0A638058/CCAPI/ccGetMediaClassTag:
media class tag 0 Jul 12 19:44:10.279: //184/4F1E0A638058/CCAPI/ccSetMediaclassIp2ipTags: media class tags set: NR 0, ASP 0 Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccGetMediaClassTag: no peer tag found, use the default media class tag Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccSetMediaclassIp2ipTags: media class tags set: NR 0, ASP 0 Jul 12 19:44:10.279: //184/4F1E0A638058/CCAPI/ccGet_xc_nr_asp_info: media class tags: NR 0, ASP 0 Jul 12 19:44:10.279: //183/4F1E0A638058/CCAPI/ccGet_xc_nr_asp_info: media class tags: NR 0, ASP 0 Jul 12 19:44:25.283: //184/4F1E0A638058/CCAPI/cc_api_call_disconnected: Cause Value=38, Interface=0x272870C, Call Id=184 Jul 12 19:44:25.283: //184/4F1E0A638058/CCAPI/cc_api_call_disconnected: Call Entry(Responsed=TRUE, Cause Value=38, Retry Count=0) Jul 12 19:44:25.283: //183/4F1E0A638058/CCAPI/ccCallReleaseResources: release reserved xcoding resource. Jul 12 19:44:25.283: //184/4F1E0A638058/CCAPI/ccCallSetAAA_Accounting: Accounting=0, Call Id=184 Jul 12 19:44:25.283: //184/4F1E0A638058/CCAPI/ccCallDisconnect: Cause Value=38, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=38) Jul 12 19:44:25.283: //184/4F1E0A638058/CCAPI/ccCallDisconnect: Cause Value=38, Call Entry(Responsed=TRUE, Cause Value=38) Jul 12 19:44:25.283: //184/4F1E0A638058/CCAPI/cc_api_get_transfer_info: Transfer Number=NULL Jul 12 19:44:25.283: //184/4F1E0A638058/CCAPI/cc_api_call_disconnect_done: Disposition=0, Interface=0x272870C, Tag=0x0, Call Id=184, Call Entry(Disconnect Cause=38, Voice Class Cause Code=0, Retry Count=0) Jul 12 19:44:25.283: //184/4F1E0A638058/CCAPI/cc_api_call_disconnect_done: Call Disconnect Event Sent Jul 12 19:44:25.283: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Jul 12 19:44:25.283: :cc_free_feature_vsa freeing 102AC100 Jul 12 19:44:25.283: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Jul 12 19:44:25.283: vsacount in free is 1 Jul 12 19:44:25.283: //183/4F1E0A638058/CCAPI/ccCallDisconnect: Cause Value=38, Tag=0x0, Call Entry(Previous Disconnect Cause=0, Disconnect Cause=0) Jul 12 19:44:25.283: //183/4F1E0A638058/CCAPI/ccCallDisconnect: Cause Value=38, Call Entry(Responsed=TRUE, Cause Value=38) Jul 12 19:44:25.283: //183/4F1E0A638058/CCAPI/cc_api_get_transfer_info: Transfer Number=NULL Jul 12 19:44:25.439: //183/4F1E0A638058/CCAPI/cc_api_call_disconnect_done:
Disposition=0, Interface=0x161B8070, Tag=0x0, Call Id=183, Call Entry(Disconnect Cause=38, Voice Class Cause Code=0, Retry Count=0) Jul 12 19:44:25.439: //183/4F1E0A638058/CCAPI/cc_api_call_disconnect_done: Call Disconnect Event Sent Jul 12 19:44:25.439: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Jul 12 19:44:25.439: :cc_free_feature_vsa freeing 102ACAA0 Jul 12 19:44:25.439: //-1/xxxxxxxxxxxx/CCAPI/cc_free_feature_vsa:
Jul 12 19:44:25.439: vsacount in free is 0
======================================================================================================
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Labels:
02-15-2016
04:00 AM
Hi Islam,
Will you be able to assist me we have installed Appspace on a VM Server we followed the guide and did everything.
But when we need to log in and I use http://localhost/app or http://myserverIP//app I get this error on the webpage. "The link you followed requires additional authorization for access" attachment_1.png
I don't know what the issue might be.
Kind Regards
... View more
Created by
corneliss
in IP Telephony and Phones
07-24-2019
07-24-2019
Hi Jinto, I will try to get this to you as RTMT takes forever to load
the messages and I need to rem...
Created by
corneliss
in IP Telephony and Phones
07-24-2019
07-24-2019
Hi Kevin, Yes it does use T.38 I did see the T.38 request coming through
when doing a test from a PR...
Created by
corneliss
in IP Telephony and Phones
07-23-2019
07-23-2019
Hi Jinto Sorry about that, let me explain quick. First Line is my CUBE
Router 192.168.1.10Second lin...
Created by
corneliss
in IP Telephony and Phones
07-23-2019
07-23-2019
Hi Jinto, Please see attached is the error from RTMT.
Created by
corneliss
in IP Telephony and Phones
07-23-2019
07-23-2019
Hi Jinto, I have tried the following you have suggested as before but
the issue always remains. Runn...
Created by
corneliss
in IP Telephony and Phones
07-22-2019
07-22-2019
Hi, So under Global Config: voice service voipip address trusted
listmode border-elementallow-connec...
Created by
corneliss
in IP Telephony and Phones
07-20-2019
07-20-2019
Hi All, I need some help, we have recently started moving from MGCP
Gateways to SIP CUBE on our Rout...
Created by
corneliss
in Unified Communications Infrastructure
06-02-2019
06-02-2019
Hey everyone, we had an issue with extension mobility that just stopped
working. A user would log in...
Created by
corneliss
in TelePresence and Video Infrastructure
05-06-2019
05-06-2019
@Anurag Srivastava1 wrote:Hello, Please give us more details about your
deployment to understand the...
04-29-2019
Hey, When changing the Web bridge settings on CMS we are able to login
using the Webrtc page and con...
Created by
corneliss
in TelePresence and Video Infrastructure
01-15-2019
01-15-2019
Hey All, We are currently implementing a Bot for scheduling Meetings via
the API Integration in TMS....
Created by
corneliss
in Collaboration Applications
10-23-2018
10-23-2018
Issue I had When doing a call using a HDX7000 running version 3.1.12 and
dialing to CMS on version 2...
Created by
corneliss
in TelePresence and Video Infrastructure
06-12-2018
06-12-2018
Hey Sagsheth, Please send me the tool as well.
Created by
corneliss
in TelePresence and Video Infrastructure
05-07-2018
05-07-2018
Hey, This is not really a problem, I just need some insight from you
guys on the best way to approac...
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