Voice over IP

Hello,My company has bought Cisco 1841 router to improve voice quality for our IP Phones as our IP Phone provider only support Cisco routers.But we are still having an issues with our IP Phone.  We have been informed by IP Phone provider that our pho...

What are some alternatives to the Cisco attendant console software. The only thing I could find was ARC, but the price was very steep. Also what is the recommended number of attendant console instances per Call Manager cluster.

hi all,i faced an issue, my pub is not accessible thru URL... but i can reach sub wt shuld i check first?any response will be much appreciated....

i have traffic on a private ip range sending sip traffic that is being natted to another range on a cisco router. how do i get the nat router to change the ip address embedded in the sip to the nat address.

I need help in making the above subject a reality. I have two sites A and B, both connected together with a Leased WAN Link. Site has a Panasonic TDA100, and site B has CUCM. For now, bith sites cannot call each other on phone extensions, they only m...

Hello,     we have a 1900 series router that is running CME and CUE. Now the end users are raising concerns that when they transfer a call from one phone to another they here 'transferring to xxxx'. on  some phone they hear the auto attendant say "tr...

Dear support,I have CUCM v6 in back office and I have VPN connection from another office and all IP Phone is connected and registered to CUCM and they have land lines connected to getaway router. Extension between two site working well. But when they...

Hi, bodiesWe have acquired some ATAs 187 (SIP) recently, and we are having problems with some VoIP calls that worked properly on ATA 186 (SCCP) we only hears a busy tone and call setup ends, we have tried almost everything without success.Thanks in ...

hello want ask if there are way to connect linksys SPA400 with  Call Manager Express  ???

Hello,I'm planning on installing a Cisco Unified Communications Manager as a subsequent node, and I'm wondering if the subsequent node needs it's own SIP trunk for outbound calls, or it uses the trunk from the primary node?Thanks,Maksim

This question relates to the "anonymous call reject" feature.  I have been advised that the CISCO VoIP system can't differentiate between an incoming call where "no caller ID information is passed" versus an incoming call where the caller has "delibe...

We a working FoIP Implementation:RightFax SR140, Intercluster-Trunk to CUCM 6.1.2, 2811 H.323 voice gateway with IOS 12.4(24)T3 to PSTN.The problem is, outbound FAX to a FAX-Machine, that answers the call only after more than 15s will be disonnected,...

Hi,Can anyone tell me how I can implement after hours call forwarding in call manager. After 5:00 PM all the calls should be forwarded to mobile phone.How can this be done in call manager????? Thanks.

Not sure if this is the right place to ask this question.Cisco IOS voice command reference for "output attenuation" and "input gain" mentions that a default attenuation and gain plan is in effect (-6DB).Is there a way to disable the output attenuatio...