frederick

When pressing * or #, the AA doesn't accept these user inputs for redirection to another AA (for example). But if I replace with a number instead of * or # it works fine, and the call is re-routed to a destination. Can you comment? Thanks again.

Hi! I think I was able to figure it out. Instead of using the IVRNode, I used the AA (IVR Tab), and I was able able to route the call to Voicemail prompt. Thanks again.

It is the IVR Node (with Account No.). From Service Flag Node to IVR Node then "voice mail-agent", how can we link the IVR to voicemail prompt? So the Caller will listen to the IVR ("thank you for calling... ), then he will be forwarded to voice mail prompt. Thanks again.

I mean from IVR Node to Voice Mail of "8"+43 (Agent#43). Where can I put the 843 in the IVR settings? Thanks again,

Hi! I'm doing some tests on re-routing based on business hours and holidays for both english and french accounts. I was able to re-route calls during holidays scenario and sent them to an IVR (language). But my question is how can I activate a mailbox and link this mailbox to the IVR? Our objective is when the Caller listens to an IVR, he can be able to leave a message on a Voice Mail (any agent's voice mail number). Example: Caller ---> French ---> Holidays IVR (En) ---> Voice MailBox Number 1 (to leave a message) Caller ---> English ---> Holidays IVR (Fr) ---> Voice MailBox Number 2 (to leave a message) Thanks again.

I just want to brought back this issue if there's a way that PBXnSIP support InBand DTMF? If yes, what configuration we can do to make it work? Thanks again.

Does this mean that if a Voip-Device communicates DTMF-Inband to our PBXnSIP server, the dtmf will not be detected? Which particular settings we can look to make it work (both rfc2833 and inband)? Thanks again.

I see in the general settings the Inband DTMF detection: ON; But when I tried to do an inbound call with Inband-DTMF, the pbxnsip cannot detect this. What could be wrong? Also I did Log Level 9 for general logging and log media events ON, but can't get the logs I would like to analyze for Inband-DTMF. When I used DTMF-rfc2833 on my device, the DTMF works! and I can get some logs (example: [6] 2009/01/16 10:02:46: Received DTMF 1 ).

Thank you. I used it in the ACD. it works fine!

Okay, we are trying to identify callers based on the selection they do in the AA or Agent Group, for ex. 1 caller selected gold account, another caller selected basic account, etc. Using a caller ID name, we can see (from the agent's caller ID display) that the call is for gold clients queue or basic queue, but the calling number has to be the original number from the Client. Thanks again.

Basically, we would like to change the inbound Caller ID name, but retain the Caller Telephone number. Is this inbound Caller ID modification supported in the AA side? If not are there any plans to include this in the future? Thanks,

Hi! unfortunately, I can only simulate few calls as my sip line has limitations of concurrent calls, but I was able to detect the problem before. Now, with the new version it works fine. I don't have an environment to simulate the scenario you are referring with.

We would like to modify the caller id per AA during inbound call to the agents. How can this be realized? I tried the ANI in the AA but didnt worked. I can still receive the Original Inbound Caller ID. Thanks.

Thank you. With your idea, yes i see that my switch rejects the outbound calls, the reason is the "from field" being sent out by pbxnsip is with some additional numbers. The Caller ID was added in the main number: example: sip.From == "<sip:1514448038243@sip.domain.ca> In the ext.43, I see that Block outgoing caller-ID: is NO, then I changed this to YES, the from field becomes; sip.From == "<sip:15144480382@sip.domain.ca> That fix my issue. But, my question now is that if this Block Outgoing Caller-ID is set to NO, why the ext. number was attached with the main number in the from field? Thanks again.

This could be easy, but I tried to check the config and still cannot find the reason why my outbound calls are always rejected by the pbx. Inboud calls are OK. I'm always getting this rejection in the logs: [5] 2008/12/18 11:31:13: Dialplan Standard User: Match 915148793318@192.168.150.43 to <sip:15148793318@sip.domain.ca;user=phone> on trunk 15144480382 [5] 2008/12/18 11:31:13: INVITE Response: Terminate 61ce2f6c@pbx Which config does this error points to? Thanks again, Frederick

We verified the simultaneous calls, it is now working! Thanks to update-3.1.1.3107;

Hi! this link is not working http://pbxnsip.com/download/pbxctrl-3.1.1.3100.exe, where can we get this version? I also tried to go to the download software area of pbxnsip but didnt find this version; I just saw the following other versions: PBX 3.0.1.3023 2008/10/03 Windows 32 Executable PBX 3.0.1.3023 2008/10/03 Windows 32 (update) Executable PBX 3.0.1.3023 2008/10/03 Mac OS (Darwin 9.0) Executable PBX 3.1.0.3043 2008/11/24 Linux (CentOS 5) Executable PBX 3.0.1.3023 2008/10/03 Linux (SuSE 10) Executable PBX 3.0.1.3023 2008/10/03 Linux (RedHat ES4) Executable PBX 3.0.1.3023 2008/10/03 Linux (Debian 4.0) Executable PBX 3.0.1.3023 2008/8/13 CS410 Update package Executable PBX 2.1.14.2498 2008/8/8 Windows 32 Executable PBX 2.1.14.2498 2008/8/8 Windows 32 (update) Executable PBX 2.1.14.2498 2008/8/8 Linux (SuSE 10) Executable PBX 2.1.14.2498 2008/8/8 Linux (RedHat ES4) Executable PBX 2.1.14.2498 2008/8/8 Linux (Debian 4.0) Executable Thanks!

Is there any defined timeline for the commercial release of this version 3.1.1?

Hi! Juan, In fact it is to the 3rd option in my case, then I change it to (2) Ring longest idle first, and also to Random (default). It didn't worked on both cases. Strange is, if the other agent (on call) hang-up, it will start ring the free agent (idle), and later on the other agent. Any other advise? Thanks, Fred

OK thank you. But when is your tentative timeline for the version 3.1.1?

Yes, the 2nd Caller on the queue just remain on the queue and just listening to music, it doesn't advanced to the agents, UNLESS the 1st Caller hang-up, then both agents will ring. Please confirm when can we realize this solution. Thanks, Fred

I have an issue on queue processing for PbxNSip; This is when I have simultaneous callers on queue, and 1 caller is picked-up, the 2nd caller continuous to be on-queue instead of ringing the free Extension. But when the 1st Caller hangs-up (or the Extension handling that call hangs-up), the call from 2nd Caller immediately went thru both Extensions. Is there a parameter setting to make simultaneous callers on queue be handled by different Extensions? That is when the 1st Caller on queue was answered, the 2nd Caller will be sent to the free Extension and be answered as well. Thanks, Fred