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Ventus Logistics

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  1. Trunk setup for correct ANI

    in General Setup

    Posted

    I think he is talking about, http://wiki.pbxnsip.com/index.php/Release_..._3.1#Cell_Phone

     

    Also, when the cell phone was used to place an outbound call, the PBX sometimes presented the Caller-ID of the cell phone, not the extension. The purpose was to hide it. That was fixed, and now users can use their cell phones to place outbound calls and present the caller-ID of the PBX.

     

    For most cases this problem still exists, I did manage to get it working with 1 ITSP, creating an inbound trunk and an outbound trunk, and some how forcing it to display the trunk ANI.

    Hello All. Sorry if i dont explain my situation clearly. let me go back to basics.

     

    I have both scenarios working perfectly with pbxnsip and my PRI provider that connect to my audiocodes mediant 1000. The problem is that I cannot mix both scenarios. If I make scenario 1 work I have the issue scenario 2 stops working and vice versa.

     

    My need is to having both scenarios at the same time work.

     

    1.) When the owner of the extension dials in to the DISA or AA (pbxnsip) from the mobile (763-1599) and dials out to the PSTN I need for the receiving party to see the pbxnsip ANI (790-1509) setup on his extension. For this to work I have to setup the remote party ID on the trunk to be P Asserted Identity.

     

    2.) When any person in the world dials from the PSTN (example 722-5026) into our pbxnsip user DID 790-1509 the call is immediatly forwarded to that user mobile # 763-1599 and we need our mobile user to be able to identify who the caller is via there caller id (722-5026). For this to work I have to setup the remote party ID on the trunk to P Prefered Identiy.

     

    As you see above both scenario 1 and 2 work great on individual cases. The problem starts when I want to have both scenarios working at the same time as we only have 1 setting available for the trunk.

     

    The solution I see is having an option to set 2 type of identities on a trunk and be able to have it use a specific type when the call is being generated via DISA or AA. If the call is just a redirect or immediate call forward or rest of the scenarios use the other type of identity.

     

    You might ask why our world is so complicated and the reason is for tricking our mobile carriers that the call is always coming from a particular number for the office users that dont have unlimited mobile plans and this saves us a lot of money on our mobile bills using there friends and family plans. For the rest of our users that do have unlimited plans or regular plans we do need them to see the original caller ID coming in.

     

    thanks for your help and I hope i was able to explain better this time.

  2. Trunk setup for correct ANI

    in General Setup

    Posted

    It is still not clear to me what the setup is. Maybe you can include a little drawing. No big art, just something scribbled on paper and scan it... B)

     

    Maybe the problem is that this is not a trunk call? ANI has no effect on calls that are to extensions.

     

    Sorry I am not big with forums or posting comments via text as it is more complicated to interpret. Is there any phone support even if it is paid so I can convey my message.

     

    My scenario is dialing out to the PSTN where in some cases we need the calling party ANI to go through pbxnsip al the way to the destination #. This happenes when a PSTN (489-1500) user dials our DID 790-1509 which on the pbx is setup as extension 1509 with immediate call forward to 7631599. We want the original calling party (489-1500) ANI showed up on mobile 7631599 and this does happen with P-Preffered Identity setup on our trunk.

     

    everything has been working great.

     

    know we have a new scenario where we have 763-1599 calling the auto attendant and being authenticated by caller id and pressing 1 to place out bound calls to PSTN. we want to hide the 763-1599 ANI from end user receiving call on PSTN. but for that to work we have to setup trunk to P-Asserted Identity so it respects the ANI of the office extension (956-790-1509). so when we do this change we loos the previous working scenario.

     

    so that is where my question comes into play to have both scenarios working what setup can we have? could you make the option on the extension setup to have P-Asserted Identity and override the trunk setup when you see this option.

  3. Trunk setup for correct ANI

    in General Setup

    Posted

    Remote-Party-ID is a old, obsolete Cisco proposal for indicating the remote party. It expired a couple of years ago. Everyone in this industry should better stop using it. We are too nice and we still offer it. I believe even Cisco does not offer it any more in the latest versions.

     

    P-Asserted-Identity and P-Preferred-Identity made it to the RFC state. This is the way to go. AudioCodes as one of the better products in this industry fully supports it.

     

    Regarding the cell phone redirection, there is the old problem: How can the gateway "trust" the PBX that this is not a spoofed caller-ID? Especially for ITSP that is a huge problem. Because of this, we introduced a new header called "Related-Call-ID" that indicates the original caller-ID. Not RFC yet, but at least CallCentric said they are going to support it and then the original caller-ID will show up on the cell phone. Bravo!

     

    Good morning. Please read my post more carefully. Remote Party ID/xxxx as I posted is also the name of the field in the trunk setup where you configure these options. I never mentioned I was using remote party ID. I am using Preferred and it works fine but it will not respect the ANI of an Extension even if you specify it under ext. config. In order to get this scenario to work of respecting your ANI config you have to use P-Asserted and then it works fine. The problem with P-Asserted is that we loose what we have working under P-Preffered which is having the original caller ID displayed on call forwarded calls.

     

    So to sum up both scenarios work correct on PBXNSIP but my need is not to have one of them working at 1 time. I need both working simulteanously so our current setup will work 100%. How can this be acheived?

  4. Trunk setup for correct ANI

    in General Setup

    Posted

    Hello. We have 7 offices with PBXNSIP on all of them and we have them interconnected. We have most of our PSTN traffic setup to go out our Audio Codes Mediant 1000 which has 2 trunks. One with a mexican carrier and another with a US carrier. Each carriers deleivers 100 DID's in both countries and they match to 1500 - 1599. We have our Remote Party/Privacy Indication: on the trunk set to P Prefered Identity in order to have our ANI's displayed correctly as we do alot of call forwarding of some of the DID's to different mobiles and land lines and we need the remote party number displayed. It has all been working great for sometime know.

     

    Just last week we came into the need of having our user dialing into calling card DID or auto attendent DID and having them dial out via the PBX for cost and control reasons. We noticed that the called party was being displayed the orginal ANI which was coming from a mobile or remote land line. In order for us to get PBXNSIP to respect the ANI of the extension being used we have change the Remote Party/Privacy Indication:on the trunk to P Asserted Identity. With this the 2nd scenario works great but we loose the first scenario.

     

    Is there any way to have more control over Remote Party/Privacy Indication: on a trunk? Maybe something where we can control via the extension/DID?

  5. Caller ID match

    in General Setup

    Posted

    Hello. We are seeing an issue with the Caller ID match when a call comes in to the auto attendent. We have our mobile phones setup on a user extension profile with a prefix infront of the number so it can fork the call to the mobile number. the problem is that when you match the caller id of the incoming call you dont recognize the number because of the prefix. if i remove the prefix it works fine for the incoming call to get authenticated but the forking stops working.

     

    all our call have either a prefix of 8 for USA calls which send to our audio codes mediant 1000 for the specific trunk to the USA and a 9 for mexico.

     

    example of mobile number 7631599 but we have setup as 87631599 on the mobile setting.

     

    i know we could do manual routes on the dial plan to add the 8 infront but then that makes us modify all our dialplans.

     

    my suggestion is to read the caller id from right to left until you match the 7 digits and that it could be configurable per the system admin.

  7. cs410 FXO ports are dead every morning

    in Embedded

    Posted

    Hello,

     

    I have 3 cs410 (black boxes) working in 3 different locations and every morning my gent have to restart each box because every morning they don't recognize neither the incoming nor the outgoing calls.

     

    My IT guy made a capture usign the comcerto:~# netstat -pan through the cs410 console and he got this:

     

    SSH Secure Shell 3.2.9 (Build 283)

    Copyright © 2000-2003 SSH Communications Security Corp - http://www.ssh.com/

     

    This copy of SSH Secure Shell is a non-commercial version.

    This version does not include PKI and PKCS #11 functionality.

     

     

    comcerto:~# netstat -pan

    Active Internet connections (servers and established)

    Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name

    tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN 260/pbxctrl-debian3

    tcp 0 0 0.0.0.0:5061 0.0.0.0:* LISTEN 260/pbxctrl-debian3

    tcp 0 0 0.0.0.0:37 0.0.0.0:* LISTEN 230/inetd

    tcp 0 0 0.0.0.0:9 0.0.0.0:* LISTEN 230/inetd

    tcp 0 0 0.0.0.0:13 0.0.0.0:* LISTEN 230/inetd

    tcp 0 0 0.0.0.0:111 0.0.0.0:* LISTEN 191/portmap

    tcp 0 0 0.0.0.0:80 0.0.0.0:* LISTEN 260/pbxctrl-debian3

    tcp 0 0 0.0.0.0:881 0.0.0.0:* LISTEN 275/rpc.statd

    tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN 269/sshd

    tcp 0 0 0.0.0.0:25 0.0.0.0:* LISTEN 230/inetd

    tcp 0 0 0.0.0.0:443 0.0.0.0:* LISTEN 260/pbxctrl-debian3

    tcp 0 0 192.168.55.210:80 192.168.50.82:2347 ESTABLISHED260/pbxctrl-debian3

    tcp 0 1008 192.168.55.210:22 192.168.52.110:4490 ESTABLISHED1224/0

    tcp 0 0 192.168.55.210:80 192.168.50.82:2348 ESTABLISHED260/pbxctrl-debian3

    udp 0 0 0.0.0.0:1024 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:1025 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:1026 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:9 0.0.0.0:* 230/inetd

    udp 0 0 0.0.0.0:161 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:5060 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:68 0.0.0.0:* 186/dhclient3

    udp 0 0 0.0.0.0:69 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:875 0.0.0.0:* 275/rpc.statd

    udp 0 0 0.0.0.0:878 0.0.0.0:* 275/rpc.statd

    udp 0 0 0.0.0.0:111 0.0.0.0:* 191/portmap

    Active UNIX domain sockets (servers and established)

    Proto RefCnt Flags Type State I-Node PID/Program name Path

    unix 5 [ ] DGRAM 784 219/syslogd /dev/log

    unix 2 [ ] DGRAM 957 186/dhclient3

    unix 2 [ ] DGRAM 891 275/rpc.statd

    unix 2 [ ] DGRAM 803 222/klogd

    comcerto:~#

    comcerto:~#

    comcerto:~#

    comcerto:~#

    comcerto:~#

    comcerto:~#

    comcerto:~# comcerto:~#

    Broadcast message from root (Fri Aug 8 22:00:08 2008):

     

    The system is going down for reboot NOW!

    comcerto:~# netstat -pan

    Active Internet connections (servers and established)

    Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name

    tcp 0 0 0.0.0.0:5060 0.0.0.0:* LISTEN 260/pbxctrl-debian3

    tcp 0 0 0.0.0.0:5061 0.0.0.0:* LISTEN 260/pbxctrl-debian3

    tcp 0 0 0.0.0.0:37 0.0.0.0:* LISTEN 230/inetd

    tcp 0 0 0.0.0.0:9 0.0.0.0:* LISTEN 230/inetd

    tcp 0 0 0.0.0.0:13 0.0.0.0:* LISTEN 230/inetd

    tcp 0 0 0.0.0.0:111 0.0.0.0:* LISTEN 191/portmap

    tcp 0 0 0.0.0.0:80 0.0.0.0:* LISTEN 260/pbxctrl-debian3

    tcp 0 0 0.0.0.0:881 0.0.0.0:* LISTEN 275/rpc.statd

    tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN 269/sshd

    tcp 0 0 0.0.0.0:25 0.0.0.0:* LISTEN 230/inetd

    tcp 0 0 0.0.0.0:443 0.0.0.0:* LISTEN 260/pbxctrl-debian3

    tcp 0 1008 192.168.55.210:22 192.168.52.110:4509 ESTABLISHED293/0

    udp 0 0 0.0.0.0:1024 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:1025 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:1026 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:9 0.0.0.0:* 230/inetd

    udp 0 0 0.0.0.0:161 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:5060 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 0.0.0.0:68 0.0.0.0:* 186/dhclient3

    udp 0 0 0.0.0.0:69 0.0.0.0:* 260/pbxctrl-debian3

    udp 0 0 127.0.0.1:5062 0.0.0.0:* 258/sipfxo

    udp 0 0 0.0.0.0:875 0.0.0.0:* 275/rpc.statd

    udp 0 0 0.0.0.0:878 0.0.0.0:* 275/rpc.statd

    udp 0 0 0.0.0.0:111 0.0.0.0:* 191/portmap

    Active UNIX domain sockets (servers and established)

    Proto RefCnt Flags Type State I-Node PID/Program name Path

    unix 4 [ ] DGRAM 784 219/syslogd /dev/log

    unix 2 [ ] DGRAM 891 275/rpc.statd

    unix 2 [ ] DGRAM 803 222/klogd

    comcerto:~#

     

    I think that there is an internal problem with the software and the cs410's GW SIP FXO. The 3 boxes are running the latest version.

     

    Please need some help here!!!

     

    Thanks,

     

    Enrique

  8. White CS410 version

    in Embedded

    Posted

    Is 3.0.0.2899 suitable or recommended to run on the White CS410s'?

    Do not upgrade to the 3.0.0.2899 version. we have 2 white boxes that we upgraded and the FXO ports are dead. everything else works but we had to buy new ones.

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