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- About Jagpreet Barmi
05-22-2019
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Member since
04-30-2013
64
Posts
135
Helpful
14
Solutions
07-15-2015
10:17 PM
Hi Kiran, Correct me if I didn't get the issue correctly. Call flow: User B (078-333-3333) calls User A (0207-111-1111) --> CFA to User A Mobile (077-222-2222). In this scenario, User B becomes the calling party, User A becomes the Redirecting Party and Mobile number becomes the called party. Under the SIP Trunk, the Calling Party Selection - should be set to 'Originator' to display the calling party as User B (078-333-3333). HTH, Jagpreet Singh Barmi
... View more
Created by
Jagpreet Barmi
in IP Telephony and Phones
in IP Telephony and Phones
01-18-2015
02:06 PM
01-18-2015
02:06 PM
Hi, It appears to be due to the following defects: https://tools.cisco.com/bugsearch/bug/CSCti14618/?reffering_site=dumpcr https://tools.cisco.com/bugsearch/bug/CSCtw46611/?reffering_site=dumpcr Workaround: Set selinux to permissive mode using "utils os secure permissive". If NTP syncs correctly and alarms stop then contact TAC for a permanent workaround. selinux is a security feature. It should only be disabled for debugging purposes. Change SELinux back to enforcing using CLI command "utils os secure enforce". To confirm, please check the ntp stratum and sync status (utils ntp status) on the affected server and make sure its stratum is below 5 and NTP is in sync. If it is good, you may wanna try the workaround mentioned in the defect above and contact TAC for its permanent fix. HTH, Jagpreet Singh Barmi
... View more
Created by
Jagpreet Barmi
in IP Telephony and Phones
in IP Telephony and Phones
01-18-2015
01:43 PM
01-18-2015
01:43 PM
Hi Vijay, Have you checked the Restriction Tables in Unity Connection? Try adding a wildcard (*) in the rules and check again. Regards, Jagpreet Singh Barmi
... View more
09-27-2014
05:24 AM
Hi Jhun, Both the calls selects the same incoming dial peer 3000 and outgoing dial peer 2010. Both the calls are sent to the same PRI to provider. Comparing the ISDN setup messages, they are similar to each other wrt calling number, called number, Plan, Type. However, we see that the provider is rejecting one of the calls with reason (Unallocated/unassigned number) and allowing the other. Successful call: ISDN setup sent. Sep 26 01:24:29 CDT: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x0908 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x2181, '5122157546' Plan:ISDN, Type:National Called Party Number i = 0xC1, '9205252' Plan:ISDN, Type:Subscriber(local) Received call proceed from the provider. Sep 26 01:24:29 CDT: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x8908 Channel ID i = 0xA98381 Exclusive, Channel 1 Sep 26 01:24:29 CDT: //246955/0092E4C3975E/CCAPI/cc_api_call_proceeding: NAUSVG01-XO#Interface=0x4224DD28, Progress Indication=NULL(0) Received Alerting from the provider. Sep 26 01:24:31 CDT: ISDN Se0/0/0:23 Q931: RX <- ALERTING pd = 8 callref = 0x8908 Progress Ind i = 0x8088 - In-band info or appropriate now available Received Connect from the provider. Sep 26 01:24:31 CDT: ISDN Se0/0/0:23 Q931: RX <- CONNECT pd = 8 callref = 0x8908 Failed call: ISDN sent sent to the provider. Sep 26 01:25:17 CDT: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x0909 Bearer Capability i = 0x8090A2 Standard = CCITT Transfer Capability = Speech Transfer Mode = Circuit Transfer Rate = 64 kbit/s Channel ID i = 0xA98381 Exclusive, Channel 1 Calling Party Number i = 0x2181, '5122157546' Plan:ISDN, Type:National Called Party Number i = 0xC1, '9829053' Plan:ISDN, Type:Subscriber(local) Received Release complete from the provider. Sep 26 01:25:17 CDT: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x8909 Cause i = 0x8081 - Unallocated/unassigned number As mentioned by Carlo, its an issue with the provider and you need to check with them. HTH, Jagpreet Singh Barmi
... View more
07-25-2014
12:05 AM
Thats Great Jim!!! :-) To confirm if the transcoder is allocated, you can run the following command on the router on which the transcoder is configured: show sccp connections It would show the remote IP addresses of both the endpoints where the rtp is terminated, with codec and port number. Basically, both the legs of the Xcoder. i.e. one toward the phone and other towards Lync. The snip that you have provided indicates that the callmanager is in the process of allocating the device and the allocation is not finished yet. You should see somthing like ResourceAllocated=1. Following which the number of available streams for the media resource would would decrease by 1. HTH, Jagpreet Singh Barmi
... View more
Created by
Jagpreet Barmi
in IP Telephony and Phones
in IP Telephony and Phones
07-24-2014
03:06 PM
07-24-2014
03:06 PM
Jim, I dont think this will work. The configuration between Phone and Xcoder should be G729, as per your requirement. (i.e. phone <--- g711ulaw ---> Xcoder <--- g729 ---> Trunk endpoint ) However, with the above configuration the region between phone and Xcoder is configured with g711. The previous scenario should work fine. JIM_TEST_RG (cisco phone) to AU_PER_REG (Xcoder) is using G729 AU_PER_REG (Xcoder) to AU_PER_REG (Trunk endpoint) is using G711 Additionally, in the trace file provided earlier, we see that the device configured with RegionB=AU_PER_REG is advertising support for Multiple codecs (G711ulaw, G711alaw, G729, G729AnnexA,G729AnnexB, G728 etc ) which is generally the phone. 14641577.010 |14:43:23.172 |AppInfo |MediaTerminationPointControl(120685)::getResourcesAllocated -- Logging RegionB=AU_PER_REG Caps and MTP/XCoder Region=AU_PER_REG Caps 14641577.011 |14:43:23.172 |AppInfo |MediaTerminationPointControl(120685)::logCapabilitiesinTrace -- MTP/XCoder Device Caps = 15 11 16 12 2 4 257 259 261 14641577.012 |14:43:23.172 |AppInfo |MediaTerminationPointControl(120685)::logCapabilitiesinTrace -- Device Caps = 4 2 11 12 15 84 10 83 82 107 14641577.013 |14:43:23.172 |AppInfo |DET-RegionsServer::matchCapabilities-- savedOption=0, PREF_LIST, regionA=AU_PER_REG regionB=AU_PER_REG latentCaps(A=0, B=0) kbps=64 However, the device configured with region JIM_TEST_RG supports limited codecs eg G722, G711Alaw, G711ulaw. 14641577.003 |14:43:23.172 |AppInfo |MediaTerminationPointControl(120685)::getResourcesAllocated -- Logging RegionA=JIM_TEST_RG Caps and MTP/XCoder Region=AU_PER_REG Caps 14641577.004 |14:43:23.172 |AppInfo |MediaTerminationPointControl(120685)::logCapabilitiesinTrace -- MTP/XCoder Device Caps = 15 11 16 12 2 4 257 259 261 14641577.005 |14:43:23.172 |AppInfo |MediaTerminationPointControl(120685)::logCapabilitiesinTrace -- Device Caps = 6 41 4 2 Please confirm if the phone is configred with JIM_TEST_RG region. And the XCoder and Trunk is configured with AU_PER_REG. In case the issue still exists, please provide the complete set of ccm traces for a test call with the calling and called party details. HTH, Jagpreet Singh Barmi
... View more
Created by
Jagpreet Barmi
in IP Telephony and Phones
in IP Telephony and Phones
07-23-2014
07:15 PM
07-23-2014
07:15 PM
Hi Jim, The transcoders fails to allocate because the region configuration. The region of the transcoder (AU_PER_REG) with the region of the device (JIM_TEST_RG) is configured with 8kbps (G729) or using the default Inter-region relation of 8kbps (G729). Please change it to 64kbps(G711) and the call should work. HTH, Jagpreet Singh Barmi
... View more
Created by
Jagpreet Barmi
in IP Telephony and Phones
in IP Telephony and Phones
07-22-2014
05:13 PM
07-22-2014
05:13 PM
Hi Julio, The debugs provided earlier shows the call attempted to route from dial peer 10, which is a POTS dial peer. However, in this working working, it selects dial peer 1, which is a VoIP dial peer. dial-peer voice 1 voip description VoiceMail destination-pattern 3001 session protocol sipv2 session target ipv4:192.168.6.253 dtmf-relay sip-notify codec g711ulaw no vad dial-peer voice 10 pots description llamadas LOCAL destination-pattern 9[1-9]...... direct-inward-dial port 0/1/0:0 forward-digits 7 Clearly, the called party is different so it uses different dial peers. So, the working debugs provided above are not relevant to our scenario. Could you please provide the following: 1. How are you communicating with the provider? PRI or FXO or SIP Trunk? 2. Output if "show isdn status"? 3. The dial peer to the provider which should be selected. Regards, Jagpreet Singh Barmi
... View more
Created by
Jagpreet Barmi
in IP Telephony and Phones
in IP Telephony and Phones
07-22-2014
02:38 AM
07-22-2014
02:38 AM
Hi Gary, You should not need to enable it under phone. Once the H323 gateway shows the requirement of using a TRP, the CUCM would negotiate media between the h323 gateway(Avaya CM) and phone via an MTP/Xcoder configured as the TRP. HTH, Jagpreet Singh Barmi
... View more
Created by
Jagpreet Barmi
in IP Telephony and Phones
in IP Telephony and Phones
07-21-2014
11:04 PM
07-21-2014
11:04 PM
Hi Gary, If you CUCM is 8.x and above, you can use the concept of Trusted Relay point. Two major parts of the configuration: 1. Enable the gateway to allocate/use a trusted relay point. This can be done on the gateway page. "User Trusted Relay Point"--> On. 2. Configure an MTP which can be used as a Trusted Relay Point. This is done by going to the MTP on the CUCM and check "Trusted Relay Point". Please make sure that the MTP's region configuration and caps should be approproate to be allocated in the call flow. HTH, Jagpreet Singh Barmi
... View more
07-21-2014
10:42 PM
Hi Julio, I think you mis-read my post earlier. Could you please run the same debugs again for a test call, along with the debugs showing interaction with the Provider. If you are using PRI, run (debug isdn q931), If you using SIP, use (debug ccsip message) --Jagpreet Singh Barmi
... View more
Created by
Jagpreet Barmi
in IP Telephony and Phones
in IP Telephony and Phones
07-21-2014
03:25 PM
07-21-2014
03:25 PM
Hi, Is the user added to the Standard CAR Admin Users user group? Looks like an issue with the roles. ===Snip from CUCM admin guide==== The Standard Unified CM Admin Users role includes no permissions beyond logging into Cisco Unified Communications Manager Administration. The administrator must add another authorization role to define the parts of the Cisco Unified Communications Manager Administration that the user can administer. The Standard CCMADMIN Administration role allows a user to access and make changes in all of Cisco Unified Communications Manager Administration. Any user can act as a CAR administrator; however, you must add the end user to the Cisco CAR Administrators User Group in Cisco CallManager Administration (Standard CAR Admin Users) Try adding the user to CAR Admin user's group as well. HTH, Jagpreet Singh Barmi
... View more
Created by
Jagpreet Barmi
in IP Telephony and Phones
in IP Telephony and Phones
07-21-2014
03:12 PM
07-21-2014
03:12 PM
Hi Julio, The debugs for the ccapi layer show that the Setup request was successful at 16:13:02 after matching the outbound dial peer 10. However, it disconnects the call at 16:13:10, approximately 8 seconds later. Jul 21 16:13:02.587: //1504/B6371AC69148/CCAPI/ccIFCallSetupRequestPrivate: SPI Call Setup Request Is Success; Interface Type=6, FlowMode=1 Jul 21 16:13:10.591: //1504/B6371AC69148/CCAPI/cc_api_call_disconnected: Cause Value=41, Interface=0x6623758C, Call Id=1504 Could you please run the same debugs again for a test call, along with the debugs showing interaction with the Provider. If you are using PRI, run (debug isdn q931), If you using SIP, use (debug ccsip message) I wanted to check if there is an issue with the provider's circuit and sending the disconnect. HTH, Jagpreet Singh Barmi
... View more
07-21-2014
12:23 PM
Hi, As mentioned by Islam.Kamal, Cisco 7960 and 7961 do not support iLBC codec. However, 7945 should support iLBC. Please test with Cisco 7945 phones. You can visit the following thread with the list of phones and supporting codec. This thread is quite old, so please refer the latest datasheet of the model to get the current codec support. However, it should give you an good overview. https://supportforums.cisco.com/discussion/10486016/what-ip-phone-models-support-ilbc-codec In case the issue is still experienced, please provide CM traces for a test call using the following steps. https://supportforums.cisco.com/document/126666/collecting-cucm-traces-cucm-862-tac-sr Also provide the status of the other issue (calling from site A to site B), throwing 503 service unavailable message. HTH, Jagpreet Singh Barmi
... View more
Created by
Jagpreet Barmi
in IP Telephony and Phones
in IP Telephony and Phones
07-19-2014
01:53 PM
07-19-2014
01:53 PM
Hi, It seems that are running into two different issues for these two different call flows. When calling from Site A to Site B, we are getting the error 503 Service Unavailable with Warning: 399 SB-PUB "Unable to find a device handler for the request received on port 59406 from 142.211.64.1" Check if you have a bind command added to the running config for the sip signalling and it should be binded to the interface which is specified in the destination's IP address of the call manager on Site B. If you have mulitple interfaces, please add the bind command under the dial peer towards Site B cucm. For the issue with calling from Site A to site C, we see that OK was received with iLBC codec from Site C. However, Site A responded back with a BYE with reason code 47. Can you please share call manager traces for the same with the calling and called party details? Also check if the phone that you are using on site A supports iLBC codec as a part of its caps. HTH, Jagpreet Singh Barmi
... View more
07-15-2015
Hi Kiran, Correct me if I didn't get the issue correctly. Call flow:User
B (078-333-3333) calls User...
01-18-2015
Hi,It appears to be due to the following
defects:https://tools.cisco.com/bugsearch/bug/CSCti14618/?r...
01-18-2015
Hi Vijay, Have you checked the Restriction Tables in Unity
Connection?Try adding a wildcard (*) in t...
09-27-2014
Hi Jhun, Both the calls selects the same incoming dial peer 3000 and
outgoing dial peer 2010.Both th...
07-25-2014
Thats Great Jim!!! :-)To confirm if the transcoder is allocated, you can
run the following command o...
07-24-2014
Jim,I dont think this will work.The configuration between Phone and
Xcoder should be G729, as per yo...
07-23-2014
Hi Jim,The transcoders fails to allocate because the region
configuration.The region of the transcod...
07-22-2014
Hi Julio,The debugs provided earlier shows the call attempted to route
from dial peer 10, which is a...
Created by
Jagpreet Barmi
in Unified Communications Infrastructure
07-22-2014
07-22-2014
Hi Nasir, Do you want to use the basic adhoc conference feature?If yes,
enable IP Voice Media Stream...
07-22-2014
Hi Gary, You should not need to enable it under phone. Once the H323
gateway shows the requirement o...
07-21-2014
Hi Gary, If you CUCM is 8.x and above, you can use the concept of
Trusted Relay point.Two major part...
07-21-2014
Hi Julio, I think you mis-read my post earlier. Could you please run the
same debugs again for a tes...
07-21-2014
Hi,Is the user added to the Standard CAR Admin Users user group?Looks
like an issue with the roles. ...
07-21-2014
Hi Julio,The debugs for the ccapi layer show that the Setup request was
successful at 16:13:02 after...
Public Statistics
| Date Registered | 04-30-2013 03:56 PM |
| Date Last Visited | 05-22-2019 12:07 AM |
| Total Messages Posted | 64 |
| Total Helpful Votes Received | 135 |
Helpful Votes From
| User | Helpful Count |
|---|---|
| 5 | |
| 5 | |
| 5 | |
| 5 | |
| 35 |
