Hello, We are currently using a Be6000 system and when we schedule a call forwarding on our phones (model 7945) to external numbers, it works perfectly when the original call comes from an internal position, but when it is an external call then this ...
My Company has about 350 7971 phones in productions right now. All are covered under Smartnet which will end July 31st. We have had numerous issues with the foot stand assembly breaking on them. It is not the footstand itself but the clip in the back...
Good evening guys,My new forum place, my first questions here.I come from a more traditional PBX world for the past several years administering Siemens HiPath4000 PBX (softswitch since latest version).Cisco UCM is quite a new concept for me, VoIP fro...
Hello All,I would like to connect a 2911 cisco router to sx2000 pbx using vwic2-2mft-t1/e1 card. The signaling protocol is used in sx2000 is DPNSS.The cisco router has E1 card, and the card is configured like this:!isdn switch-type primary-dpnss!cont...
I have a requirement for calls between Japan and Malaysia using a centralized CallManager MTP resource in Singapore. There is UCM servers in Japan and Singapore but not in Malaysia. Malaysia phones are using Singapore UCM as IP PBX.I have restriction...
Hi,We have implement IVR on our voice gateway 2921. It should work with incoming calls from PSTN through the SP's SIP-trunk. But the SP sends DTMF in band within g711. All calls to IVR are going through gateway to CCM and it forwards them back to gat...
Hi, We are facing problem with hearing ring back over PSTN. PSTN PRI----->MGCP GW. When user receives a inbound call over PSTN, the calling party hears busy tone, but the IP Phone/analog is not actually busy. This is happening with internal calls ...
I also posted this question on learningnetwork.cisco.com, but there aren't any good answers on that forum. Hopefully, there will have someone here who can help.The topology is basic and simple:Switch - Phone - PC--------------------------------------...
Hi All,Can't figure this one out for a while. How do you put a SIP endpoint namely the 9971 into power save mode in a CME system (ie turn off the screen when not in use).For a SCCP endpoint there are commands under "telephony-service" to do so, but ...
Dear all,does anybody can tell me which side the problem is telco or mine who is disconnecting the call. i am trying to make a international call by using access code as 97 but i can not hear any ring on the other side but the timer starts on the ph...
I have the followin cenario:SIP Client ----- Asterisk ---- Opensips ---- CUCM ---- CIPC.The call goes from CIPC to the CIP Client, but on the opposite way I got the message:80.488858 10.128.0.10 -> 10.124.2.35 SIP/SDP Request: INVITE sip:3940@10.12...
Sorry about this question on very legacy stuff, but part of my company still running Unity 5 (not unified, just regular standalone Exchange) and we can't purchase more voice mail licenses according to our global vendor. I tried escalating as it seem ...
My current provider sends 10 digits and we are migrating to a new provider that will only send 7 digits. I want to do make the migration easier by adjusting the dial peer in the gateway so that it sends the same digits to the Call Manager that are c...
I want a dial-peer destination patter to match all numbers. How do I do that? For example, I currently have:dial-peer voice 6 voip description Inbound SIP answer-address 5551234567 session protocol sipv2 session target ipv4:10.3.20.10 incoming called...
Hi all!I plan to execute an upgrade of CUCM of a cluster from version 7.1.5 (three MCS7835 servers) on version 9.1 (on two UCS C260 TRC servers). On the basis of documentation and requirements of the Customer to breaks in telephony service I ...
I am trying to connect my 2851 router CME 8.6 to a service using the GSM codec. The version they are using is The codec is called "GSM FR" and is the 6.10 standard. The SDP line for it is "a=rtpmap:3 GSM/8000" but when I select GSM on the fouter in ...
