About sherif.elella

sherif.elella · 04-18-2017

dear community, i am routing a call as such: phone --> CUCM X--> SIP Trunk A --> CUCME --> SIP Trunk A --> CUCM X --> SIP Trunk B --> non-Cisco SIP Server when i initiate a call, it shows me under "show sip-ua calls" two RTP Streams that dont make...

sherif.elella · 04-06-2017

here is the topology (non-cisco SIP server) <----> (sip trunk on CUCM with "mtp required" checked and "early offer" check in SIP profile) <---> CUCM <---> (SIP Trunk to CME) <--> Phone with SNR configured call flow: 1. Cisco phone registered to CU...

sherif.elella · 02-28-2017

here is my CME Config voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip sip bind control source-interface GigabitEthernet0/0 bind media source-interface Gigabit...

sherif.elella · 01-26-2017

fancy a puzzle? i am trying to integrate a non-cisco with call manager, and i added the mac address using an advanced SIP Client for the multiple extensions feature. the support for that non-cisco device claimed that he worked on another project whe...

sherif.elella · 10-16-2016

hi people, uccx 9.1, i have a team and a CSQ setup, working fine no issues. when i create a new team with a new CSQ and my resource is assigned to the new CSQ and part of the new team, its shows on the supervisor under the first team, i closed the...

sherif.elella · 04-09-2017

I tested the same call again and it worked, opened sh sccp conn on the MTP VG and sh voip rtp conn det the call flow is good, a transcoder is invoked, not an MTP XX#sh sccp conn sess_id conn_id stype mode codec ripaddr rport sport 16982270 34419...

sherif.elella · 04-06-2017

10.109.11.2 is CME 10.10.10.14 is a MTP on a voice gateway 10.10.97.106 is the cucm also MTP

sherif.elella · 04-06-2017

the calls between the non cisco server and the cucm wouldn't work otherwise ((excluding the CME)) in the sdl its the phone call going from 7438 to 3266 and 7438 to 123453266

sherif.elella · 04-06-2017

already did that, the destination of RTP is correct and there is RTP being generated as long as the phone call is up, the destination of the RTP Packets are the MTP.

sherif.elella · 04-06-2017

added a voice class codec and worked, thank you. have another problem now.

Member Since	‎12-17-2015 04:01 AM
Date Last Visited	‎05-29-2019 12:06 AM
Posts	35

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DTMF & Negotiation

One way Audio in SIP to SIP Call

SNR on CME going to a call manager

SIP Client Multiple extensions

UCCX and resources

I tested the same call again

10.109.11.2 is CME

the calls between the non

already did that, the

added a voice class codec and