Fax machine issue Cisco call manager

Hi All
Im having issue trying to get a fax machine to talk to a server but im having issues and I need to find out where the Fax is configured on GUI in the Callmanager. Please could some one help.
Thanks

Thank you,
I will try them tomorrow, but I thought restarting the servers will also reset the replication
Btw, there was a time difference on the servers due to NTP failure and I have just noticed that Pub can not see one of the Subs, the replication status is 0 and
g_cucs01_ccm6_1_2_1000_13 2 Active Connected 0 Sep 15 10:21:37
g_cucs02_ccm6_1_2_1000_13 11 Active Connected 0 Sep 15 10:35:19
g_cucs03_ccm6_1_2_1000_13 19 Active Dropped 92939 Aug 20 07:51:14
g_cucs08_ccm6_1_2_1000_13 18 Active Local 0

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FAX Machine like using Call Manager Express

Hi to all,
Let me share to your my problem, before we have this analog phone which connect to the our printer to have this fax machine to send fax letters, but since the phone line we used to send fax has been cut, so my boss wants me to set up a fax service using the extension number that we have in our CME (Call Manager Express) VOIP. can any one help me how to configure and set up one extension number of the VOIP using the Call manager Express Gui v7.1 or using the CLI. since i only have a basic knowledge in networking.
This topic first appeared in the Spiceworks Community

Much of Cisco's early SIP work involved proof of concept-proving that SIP worked in an IP telephony environment, with traditional features such as caller ID, call hold, call transfer, and three-way calling. Capabilities such as announcing a call transfer to the recipient before connecting the call (which involves setting up a media stream prior to connecting the call) had to be tested to make sure they would work through RTP streams and SIP-to-public switched telephone network (PSTN) gateways. Similarly, TDM out-of-band signaling (which passes information such as caller ID data) had to be passed via the SIP network.
Most of the early adopters of SIP technology were young service provider companies looking to differentiate themselves by providing IP-based services. They include Vonage, the IP-based telecommunications services provider that used Cisco SIP gateways and SIP analog telephone adapters (ATAs) to launch its service in 2001, and B2, a Swedish IP-based telephony service provider. But even companies such as Microsoft saw the promise of SIP as far back as 2001, when it launched the SIP-enabled Windows Messenger Update for its MSN .Net initiative. The Windows Messenger Update enabled MSN users to make SIP-enabled voice over IP (VoIP) calls from the MSN Messenger client, with Cisco providing the SIP infrastructure to Microsoft service provider partners.
Try URL:
http://www.cisco.com/en/US/netsol/ns340/ns394/ns165/networking_solutions_white_paper0900aecd80131325.shtml

Connectivity Issue between ASA 5520 firewall and Cisco Call Manager

Recently i have installed ASA 5520 firewall, Below is the detail for my network
ASA 5520 inside ip: 10.12.10.2/24
Cisco Switch 3560 IP: 10.12.10.1/24 for Data and 10.12.110.2/24 for Voice
Cisco Call Manager 3825 IP: 10.12.110.2/24
The users and the IP phone are getting IP from the DHCP server which configured on cisco 3560 Switch.
the Default Gateway for Data user is 10.12.10.2/24 and
for the voice users is 10.12.110.2/24
now the problem is that the users is not able to ping 10.12.110.2 call manager. please if somebody can help in this regard. i will appreciate the prompt response against this issues.

Actually i don't wana to insert new subnet and complicate the nework. i need a simple way to solve the problem. below is the details for the asa 5520 config.
ASA Version 8.2(1)
name x.x.x.x Mobily
interface GigabitEthernet0/0
nameif inside
security-level 99
ip address 10.12.10.2 255.255.255.0
interface GigabitEthernet0/1
nameif outside
security-level 0
ip address x.x.x.x 255.255.255.252
object-group service DM_INLINE_SERVICE_1
service-object tcp-udp
service-object ip
service-object icmp
service-object udp
service-object tcp eq ftp
service-object tcp eq www
service-object tcp eq https
service-object tcp eq ssh
service-object tcp eq telnet
access-list RA_VPN_splitTunnelAcl_1 standard permit Inside-Network 255.255.255.0
access-list RA_VPN_splitTunnelAcl standard permit Inside-Network 255.255.255.0
access-list inside_nat0_outbound extended permit ip Inside-Network 255.255.255.0 10.12.10.16 255.255.255.240
access-list inside_nat0_outbound extended permit object-group DM_INLINE_SERVICE_1 10.12.10.16 255.255.255.240 Inside-Network 255.255.255.0
access-list inside_nat0_outbound_1 extended permit ip Inside-Network 255.255.255.0 10.12.10.16 255.255.255.240
pager lines 24
logging enable
logging asdm informational
mtu inside 1500
mtu outside 1500
mtu mgmt 1500
ip local pool VPN-Pool 172.16.1.1-172.16.1.30 mask 255.255.255.0
ip local pool VPN-Users 10.12.10.21-10.12.10.30 mask 255.255.255.0
no failover
icmp unreachable rate-limit 1 burst-size 1
asdm image disk0:/asdm-641.bin
asdm history enable
arp timeout 14400
global (inside) 2 interface
global (outside) 1 interface
nat (inside) 0 access-list inside_nat0_outbound_1
nat (inside) 1 Inside-Network 255.255.255.0
route outside 0.0.0.0 0.0.0.0 Mobily 1
timeout xlate 3:00:00
timeout conn 1:00:00 half-closed 0:10:00 udp 0:02:00 icmp 0:00:02
timeout sunrpc 0:10:00 h323 0:05:00 h225 1:00:00 mgcp 0:05:00 mgcp-pat 0:05:00
timeout sip 0:30:00 sip_media 0:02:00 sip-invite 0:03:00 sip-disconnect 0:02:00
timeout sip-provisional-media 0:02:00 uauth 0:05:00 absolute
timeout tcp-proxy-reassembly 0:01:00
dynamic-access-policy-record DfltAccessPolicy
http server enable
http Mgmt-Network 255.255.255.0 mgmt
http Inside-Network 255.255.255.0 inside
no snmp-server location
no snmp-server contact
snmp-server enable traps snmp authentication linkup linkdown coldstart
crypto ipsec transform-set ESP-AES-256-MD5 esp-aes-256 esp-md5-hmac
crypto ipsec transform-set ESP-DES-SHA esp-des esp-sha-hmac
crypto ipsec transform-set ESP-DES-MD5 esp-des esp-md5-hmac
crypto ipsec transform-set ESP-AES-192-MD5 esp-aes-192 esp-md5-hmac
crypto ipsec transform-set ESP-3DES-MD5 esp-3des esp-md5-hmac
crypto ipsec transform-set ESP-AES-256-SHA esp-aes-256 esp-sha-hmac
crypto ipsec transform-set ESP-AES-128-SHA esp-aes esp-sha-hmac
crypto ipsec transform-set ESP-AES-192-SHA esp-aes-192 esp-sha-hmac
crypto ipsec transform-set ESP-AES-128-MD5 esp-aes esp-md5-hmac
crypto ipsec transform-set ESP-3DES-SHA esp-3des esp-sha-hmac
crypto ipsec security-association lifetime seconds 28800
crypto ipsec security-association lifetime kilobytes 4608000
crypto dynamic-map SYSTEM_DEFAULT_CRYPTO_MAP 65535 set pfs
crypto dynamic-map SYSTEM_DEFAULT_CRYPTO_MAP 65535 set transform-set ESP-AES-128-SHA ESP-AES-128-MD5 ESP-AES-192-SHA ESP-AES-192-MD5 ESP-AES-256-SHA ESP-AES-256-MD5 ESP-3DES-SHA ESP-3DES-MD5 ESP-DES-SHA ESP-DES-MD5
crypto map outside_map 65535 ipsec-isakmp dynamic SYSTEM_DEFAULT_CRYPTO_MAP
crypto map outside_map interface outside
crypto isakmp enable outside
crypto isakmp policy 10
authentication pre-share
encryption 3des
hash sha
group 2
lifetime 86400
crypto isakmp policy 30
authentication pre-share
encryption 3des
hash md5
group 2
lifetime 86400
telnet Inside-Network 255.255.255.0 inside
telnet timeout 5
ssh Inside-Network 255.255.255.255 inside
<--- More ---> ssh timeout 5
console timeout 0
threat-detection basic-threat
threat-detection statistics access-list
no threat-detection statistics tcp-intercept
webvpn
group-policy RA_VPN internal
group-policy RA_VPN attributes
dns-server value 86.51.34.17 8.8.8.8
vpn-tunnel-protocol IPSec
split-tunnel-policy tunnelspecified
split-tunnel-network-list value RA_VPN_splitTunnelAcl
username admin password LPtK/u1LnvHTA2vO encrypted privilege 15
tunnel-group RA_VPN type remote-access
tunnel-group RA_VPN general-attributes
address-pool VPN-Users
default-group-policy RA_VPN
tunnel-group RA_VPN ipsec-attributes
pre-shared-key *
class-map inspection_default
match default-inspection-traffic
policy-map type inspect dns preset_dns_map
parameters
message-length maximum 512
policy-map global_policy
class inspection_default
inspect dns preset_dns_map
inspect ftp
inspect h323 h225
inspect h323 ras
inspect netbios
inspect rsh
inspect rtsp
inspect skinny
inspect esmtp
inspect sqlnet
inspect sunrpc
inspect tftp
inspect sip
inspect xdmcp
service-policy global_policy global
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Cryptochecksum:e5a64fa92ae465cd7dabd01ce605307d
: end

Cisco Call Manager replication issue

Hello
We are running a cluster of 3 servers with Cisco Call Manager version 6
I have noticed that the replication status is 3 on all servers and from the CM Database Status I can see queue of updates are stuck
I have restarted Pub and one Sub but the status is still 3 and queue status is still the same
Is there anything I can do to make it work?
Thank you

Thank you,
I will try them tomorrow, but I thought restarting the servers will also reset the replication
Btw, there was a time difference on the servers due to NTP failure and I have just noticed that Pub can not see one of the Subs, the replication status is 0 and
g_cucs01_ccm6_1_2_1000_13 2 Active Connected 0 Sep 15 10:21:37
g_cucs02_ccm6_1_2_1000_13 11 Active Connected 0 Sep 15 10:35:19
g_cucs03_ccm6_1_2_1000_13 19 Active Dropped 92939 Aug 20 07:51:14
g_cucs08_ccm6_1_2_1000_13 18 Active Local 0

Cisco Call Manager Administration Page Drop down menu not working

I'm running the mobile app browser on my Xoom tablet and when I log into Cisco Call Manager Administration page I get the menu's but it will not pull down the options. The same issue happens in google chrome. I use your browser on my desktop and have no issues logging into Cisco Call Manager Administration page and getting the drop down menu's to appear but the mobile version of your browser is not working. Is there a work around? I tried changing the UAString to Desktop, iPad etc and the issue is still there.

Thanks but I've tried all that. It happens in Chrome also. I've tried Opera and it works on that. I've read other forums and it looks like a Java script issue that Chrome and now Firefox Mobile cannot seem to resolve. My preference is Firefox and I'm sure as more tablets get out then this will be solved at some point. I just don't know how it is prioritized. I do know that in order for Androids to be viable in the business sector they need to address issues like these with a higher priority.

E1 no voice on Cisco Call Manager version 6

Hello,
We are running Cisco Call Manager with MGCP and E1 connection to a Siemens PSTN
We have started experiencing no voice when calling to Siemens or calling from Siemens to Cisco. The call goes through but there is no ring tone and no voice when the call is answered.
Also, I noticed that MGCP fails back to our Sub sometime, but I am not sure if these issues are somehow related
How can I troubleshoot voice problems with E1?
Thank you

Hi fgasimzade,
Can you check with Siemen what channel order do they follow.
Cisco VG follows descending order for outbound calls and expect ascending for incoming calls.
You can toggle config using below commands
interface SerialX/Y/Z:15
isdn bchan-number-order {ascending | descending}.
HTH,
Regards,
Mohammed Noor

Hi Team, I wuold like to know if you have any app to make Firefox OS working with cisco Call Manager 10.5. Something like Cisco Jabber for Android or iOS.

I'm interesting on buying a Firefox Smart Phone, but
I would like to know if are any app to install on Firefox OS smart phone in order to work with cisco call manager 10.5.
Something like Cisco Jabber for Android o iOS.
Thanks,

Hi Itech,
If Cisco Jabber has a webapp, or mobile version of their website available, you should technically be able to access it through Firefox OS.
You may also search Firefox Marketplace for an alternative solution:
* [https://marketplace.firefox.com/]
- Ralph

Starting a Cisco call manager express

I have a cisco 2651 with Cisco call manager express.
I have too 15 IP telephones 7902G
The configuration is in factory default.
Ì have to create an IP pbx and extensions of a municipality with that components.
I's like that anybody helps me in the starting configuration, Where can I find good documentation about a tipical case like this?
I have a lot experience in networking and IP telephony but I'm newbie in CCME.
Regards

To begin the configuration use "telephony-service setup" command which will prompt you for some of the basic configuration ie how many phones, starting DN, etc. To add additional feature refer to the admin guide provided by Jorge link.
Chris

Is it possible to integrate Nokia E62 with Cisco Call Manager 4.x?

The Nokia E-61 can be integrated with Cisco Call manager 4.x using a Nokia Call Connect 1.0 client.
1.Is it possible to do the same with Nokia E62.If it can be done,what would be the requirements?
2.Is Nokia E-62 PDA a Cisco Compatible Extension device?

Thanks a lot for the prompt reply.You are right in saying that nokia and cisco are increasing the number of models supported.Infact the E65 has already joined the club.But my point of interest is the nokia E62 in particular.This Nokia E62 device
1) does not support WLAN interface(while E61,E65 does).
2) It is not Cisco Commpatible extension device.(I did not find E62 in the list of Cisco Compatible extensions.chk out the attachments for more info).
3)It is not compatible with Nokia Intellisync CAll Connect 1.0(while E61 and E65 are compatible.chk out the attachment for more info).
I found this rather weird and i wondered if Nokia E62 can be integrated with Cisco Call manager ,especially a device which is not Cisco compatible.The only possibility i can see from your answer is that the Nokia E62 might not be SIP enabled(correct me if i am wrong here!)
My objective is to send a message from a CUAE script to a nokia E62 device.Please excuse me if my posting is not in the right place.i would be highly grateful to you if you could give me pointers to an appropiate location where i can get my queries answered.

Cisco Call Manager and MGCP Question

Hello,
I appreciate if somebody can help.
Scenario:
Site 1 PSTN E1----VG----Call Manager----VG--- PABX---Site 2 PSTN1 E1
I have configured a dialing pattern on Cisco call manager 6.xxxxxx to Send to VG on Site1
Both VG routers are using MGCP with call Manager.
The problem if from Site 2 tries to call 6xxxxxx the call manager is not routing the call to the VG in site 1.
I did debug ccapi inout and on Site 2 VG the call response was the number unassigned. This means that the call Manager is searching the directory for the destination but it is not searching the route patterns.
Any ideas to override this and ask the call manager to check it's destination pattern?
Thanks,

Problem solved. The VG in Site 2 was in a CSS that is not allowed to dial PSTN.
Regards,

Cisco Call Manager free license for training purpose.

I am going to setup a Cisco IP-Telephoney lab & looking for Cisco Call Manager free software license (What is Cisco has authorized for this purpose).
Is there any available?
Please help me.

Duplicate thread of
https://supportforums.cisco.com/thread/2181741
Do not SPAM the forums.

Cisco Call Manager 8.5 and Lync 2013

Dear all,
Our customer wants to integrate their Lync 2013 to the Cisco Call Manager 8.5.
Out of some not so recent information i thought that if u want to use the enterprise voice of lync you'll have to make a sip trunk between Cisco Call Manager & Lync 2013.
I think that this still counts, i've also heard that there have been a moderate amount of changes between cucm 8.5 and cucm 8.6 on matters of sip.
I'm unable to find them though, also what does CUCI - Lync do?
Kr,
Yannick Vranckx

Hi Yannick,
There can be three types of integrations between Lync and Cisco Systems
1) Direct SIP Trunk between Lync Server and Call Manager - In this, we can setup shared line between Cisco extensions and Lync extensions, and Sip trunk between the servers. Any Cisco phone / Lync client can call each other vice-versa. Initial setup needs planning, but later it does not need much configurations in client side. The following document explains the configurations needed.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucilync/9/CUCI_BK_C0B36AC1_00_cisco-uc-integration-for-microsoft.pdf
2) CUCILYNC, which is like Cisco Jabber client but connects with Lync and Call Manager: Lync Client <----> CUCILYNC <-----> CUCM
In this option, a plug-in is installed in each client side. Although initial configurations is easier, each user needs to install the client and operate it. The below document explains the configuration and setup.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucilync/9/CUCI_BK_C0545A41_00_cisco-uc-integration-for-microsoft.html
3) RCC (Remote Call Plugin) which just provides basic call control feature but for that you need Cisco Unified Presence Server: Lync Client <----> CUPS <-----> CUCM
HTH
Manish

Cisco Call Manager TFTP requests.

I was looking to see if there was a way to look at what devices request config files from a Cisco Call Manager TFTP server.
I know that you can use RTMT to look at the total number of requests made, but is there a way to know what device make the request.
Thanks in advance,
Robert

You could do this by doing a packet capture on the server as described in the link below
https://supportforums.cisco.com/docs/DOC-11599

Cisco Call Manager MIB and OID.

Hi,
One of our customer has developed their own NMS software to manage multiple OEM products.
They want to integrate Cisco Call Manager 7.0 and Cisco Router with the NMS software for monitoring/managing of Voice traffic.
The monitoring of CUCM 7.0 will consists of the following-
1. Call History
2. Missed calls.
3. Calls in progress,
4, Dropped Calls.
The Monitoring of Cisco Router will Consists of Interface statistics and Portwise Traffic statistics of LAN and WAN Ports.
I believe this monitoring is done through Call Detailed Record and if I will load the CDR MIBs in NMS software it will be able to log CRD logs from CUCM 7.0.
Please suggest that if the same is possible.??.
kindly help me in getting the MIBs/OIDs for the same.
Best regards,
Puneet K.

Hi,
For 200+ users you should better go with Cisco CallManager (CUCM). Although ISR G2 supports 250+ users, it will not provide clustering mechanism and additional HA capabilities as CUCM does.
You can read the Call Processing chapter of the CUCM SRND:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/9x/callpros.html
Hope to help, please remember to rate helpful posts.
Kind regards,
- Adrian.

Cisco Call Manager and LifeSize Endpoints

Hello dear support community,
last week I asked which version of Cisco Call Manager supports BFCP (https://supportforums.cisco.com/message/3966334).
Unfortunately I asked the wrong question.
As it turns out I just didn't need to know which version of Call Manager supports BFCP, which I know now, I also need to know if Call Manager is compatible with LifeSize systems?
Here is the setting, so you know what I'm dealing with:
The costumer uses Cisco Call Manager (Ver. 9.1) as SIP registrar for his LifeSize endpoints. The problem is, when he is making a video conference (via SIP) and he wants to share a presentation, the other side either sees him or the presentation, but not both.
We tried it with our Call Manager (Ver. 9.0) and some LifeSize endpoints. As soon as the LifeSize endpoint uses CUCM as SIP registrar, the option to share the presentation is completely gone.
I guess the costumer switched sources from cam to pc. That would explain why he just sees either video or presentation. But, as you might have guessed, it's not an acceptable solution for the costumer.
And yes, we made sure that BFCP is enabled for the endpoints.
So, what I'm asking myself and you is:
Is this known? And more important, is there a reliable workaround?
Thanks a lot in advance.
Best regards
Tobias

I've been down this road and when I asked the question Lifesize's stance was that it wasn't supported by THEM. I pointed out that CUCM now supports line-side BFCP but they essentially shrugged their shoulders. Either they are blatently disabling content sharing by virtue of using a SIP Proxy or they have a priorietary SDP/header that CUCM isn't passing through.
Either you can call Lifesize and shake them up a bit or you can wireshark the SIP dialog to see what the codec asks CUCM before disabling the sharing button. Once you know that you might be able to use SIP transparancy to pass the header through.

Fax machine issue Cisco call manager

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