Rameshkumar

Accept Redirect is already enabled in the trunk. Below is the trunk settings. #Trunk 2 aadr: analog: false bcp: behind_nat: false cid_update: cobusy: 500 Line Unavailable codec_lock: false codecs: codest: cur: dial_extension: dialplan: 2 dir: dis: false domain: 6 dtmf: false dtmf_mode: earlymedia: false expires: 3600 failover: never fraction: 128 from_source: pai from_user: glob: global: false hcv: hd: hf: {from} hpai: {trunk} hppi: hpr: {if clip true}id{fi clip true} hrpi: hru: {request-uri} ht: {to} icid: ignore_18x_sdp: false interoffice: false minimum: 10 minor: name: Exchange outbound_proxy: sip:173.180.98.133:5060;transport=tcp pcap: false prack: true prefix: redirect: true reg_account: reg_display: reg_keep: reg_registrar: 173.180.98.133 reg_user: remote_party: request_timeout: require: rfcrtp: false ring180: false rtcpxr: false rtp_begin: rtp_end: send_email: sip_port: status: tel: true trusted: false type: gateway use_epid: false use_history: false use_uuid: false user_defined_hdr: uuid: c8521892-9803-4b90-8c09-f8a0b482a62e wrtc_dest_name: wrtc_dest_number:

I've integrated almost. Only thing when I am calling to exchange I am getting 302 Moved Temporarily error. I think it is because exchange sending back re-invite on TCP port 5062. Please advise. SIP/2.0 302 Moved Temporarily FROM: "210"<sip:2128513569@test.ramesh.net;user=phone>;tag=34533 TO: <sip:2128513569@test.ramesh.net;user=phone>;epid=DAF82C1CD2;tag=9a7665d8ef CSEQ: 12935 INVITE CALL-ID: 6346f572@pbx VIA: SIP/2.0/TCP 173.115.23.185:53009;branch=z9hG4bK-12735598d07832221e3aefd2df0f01f9;rport CONTACT: <sip:7208@MAIL.vicial.internal:5062;user=phone;transport=Tcp> CONTENT-LENGTH: 0 SERVER: RTCC/5.0.0.0 MSExchangeUM/15.01.0845.039

Sorry I did not get you.

This is just to sync the contacts from outlook. How can I integrate with unified Microsoft exchange? I believe we need to create a trunk with exchange IP and and on exchange user need to be enabled for Unified Messaging? I need complete procedure to do it.

Any update please

Yes exactly.

Hi Gyus, I am using vodia 60.0.3. I want to integrate it with microsoft unified exchange server 2016. Can you let me know where I start? Or is there any relating document? I've installed PBX and exchange server on two different machines with public IP. Please advise.

Hi I am using Vodia 60.0.3. I do not see ringtones.xml under domain level customization though it's available on global pnp template. In version 5 it appears at both global and domain level. I want to make changes on ringtones.xml on domain level. Please advise.

Any update please

The local IP's for one office is: 192.168.1.x, second office 192.168.2.x, 3rd office is: 192.168.5.x

Okay I was IP routing List but honestly didn't get concept. The multicast IP is 224.1.2.111:4000 and PBX IP is 216.25.18.116. What and how I need to add things in routing.

How can I configure routing. Can you please put any example or detail. Thanks

Is it necessary we should have phones behind NAT to work. As phones are not behind NAT in any office but it work in one office.

No phones are not behind NAT.

You mean we should enable the NAT on each phone?

Hi, I am using Vodia 60.0. I've a question. Does multicast only working on the LAN? I've 3 offices and each office has 10 Yealink Phone. Configuration is same on each phone. When one office send a multicast page it only can be hear into the same office. Other 2 offices can not hear anything.

Hi, I am using Vodia 5.3.2 version. I want to send the Extension SIP password to user's via welcome email. How can I achieve it? In "email_welcome.htm" template the variable {ssi htmvar password} uses to send web password. What would be the variable to send SIP password. Thank You

Hi, Any update on this matter?

Already tried all those options but noting worked. But this should have a solution. For one user we can't restart whole server. Also after how much time that process will be killed?

I know it use to cancel/kill the process, but it's not working. If I click on it it does not do anything and call still appear in active calls.

Hi, I am using Vodia version 60. Sometimes call stuck in Active calls while actually it has been disconnected. This cause problem if user is the part of agent group, it considered it busy and can not take calls. How can I kill the process like this without restarting the service?

Hi, I am using Vodia version 60. I do not see any overview Agents on a agent group. It shows on version 5. Please advise.

I do not want to upgrade at this moment. Is there a way as soon I change the name on extension provisioning should automatically open for 30-60 seconds?

If it is why it's not working? Is there a way I can add?

Hi All, I am using Vodia 60.0 version. Before Yealink was manufacturing phone with MAC ID's 001565xxxxxx. These phones were sinking with the PBX in real time. Suppose I make a name change on extension on the PBX it replicate on the phone at the same time. Now Yealink manufacturing with MAC ID's 805EC004xxxx. These phones are not sinking with the PBX as 001565xxxxxx MAC range. On 805EC004xxxx range if I need to change name first I need to "Open account for MAC-based provisioning" then change the name and then it will replicate on phone, If I do without open provisioning it does nothing. With 001565xxxxxx range it works perfect without open provisioning. Please advise.