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Call drops after lifting up handset

sweetsaif999
Level 1
Level 1

Problem is when we call from Branch A to other branchB , after dialing the extension number, phone rings  and after lifting up the handset the call is droping with 4 or 5 beeps , and  from Branch B   the calls  can be made easily and there is no issue, we r using sip trunk its a single site depoyment model with cucm 7 and having redundant cucm on the other site  please give some hints how to troubleshoot this problems ..

7 Replies 7

linuxchild
Level 1
Level 1

I would say codec pb ,

just a question to clarify the issue , your voice gateway are configured as sip or as MGCP ?? the trunk is to the provider or between the gateways ??

Regards

Thanks for your reply we are using H.323 gateway with sip pstn trunk to the service provider

mine is like single site deployment model

site 1(Active)

Cucm 7

publisher

subscriber

H.323 gateway

600 ip phones registered to this site

site 2(redundent)

cucum 7

publisher

subscriber

H.323 gateway

150 ip phones registered at this site

branches around 4  this branches ip phones are also registered with site 1 cucm so when one of the branch ip phone wants to call to the site 2 ip phone the phone is ringing and when the reciver is pickup the call is getting drop after 3 to 4 seconds and from the site 2 calls can be received to this branch phone..and also one more question

question 2 ? i want to register the branch phones to the site2 cucm as the site1 one is active so all the phones are getting registered to this site how to force the phones to register with the site2 cucm publisher

Hi,

Are you sure about yor design.

A Publisher at each site infers that both sdites are active as you put it.

Therefore the link between the sites shoild be an intercluster trunk (H225 or SIP)

Anyway the dropping callissue sounds more like an IP routing issue

The CUCMS can signal the phones etc but there is no IP path between the phones (inter-site) therefore may be no RTP.

Hope this helps

Alex

Regards, Alex. Please rate useful posts.

I agree with the first post it might be a codec issue

but i dont think it is routing issue ( but it could be ) as with routing issue normally the call stay up with silence

saif do you have diagram describe you topology ? just want to understand your topology and call flow better to help you

Regards,

check the fast start for inbound calls and create a MTP on the vgw

Thanks all for your earliest replies actually we are using single codec g 711 and defualt device pool only so now we are integrating the other branches with this setup so this is the problem i am facing now

Question 2 ? i want to register the branch phones to the site2 cucm as the site1 one is active so all the phones are getting registered to this site1 how to force the phones to register with the site2 cucm publisher is it like while doing factory reset 123456789*0# and then we have to assign the Ip Add of the site2 Pub manually is this right procedure.

if you are using DHCP with option 150 (CUCM TFTP IP )

then provide the IP address of the TFTP server ( where the TFTP service enabled ) of cluster 2 then the phone will be redirected to cluster 2 to register there

HTH

if helpful Rate