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Update: Asterisk is working now!

Mitul was in the office today and we finally got our Tata Indicom ISDN PRI line working with Digium TE110P card. Mitul posted an entry about it explaining the cable crimping method. It was exciting to dial into the system and hear the greeting. And then to explore the directory and voice mail services. We could dial out too easily and it was working like a charm. Now we need to remove the noise from local SIP connections. When we dial out, or somebody calls in, there is no noise. The Digium card seems to be doing the magic of noise reduction, we need to find something like that in a software now.

Also got the Sify cable connection working over Linux. It was very surprising to see that they had a Mac and Linux dialer available on their site. And it worked very well.

The Cisco IP phone couldn’t configure itself with our network 😉 Need to do some more research on that tomorrow.

And yes, the genius moment of the day! We were discussing about LDAP integration of Asterisk with Mitul and he said why don’t you use MySQL? Even for PAM?

Did some research on that, and looks like pam_mysql is stable. So we will be moving all our authentication to MySQL now. (there go the sleepless nights spent on LDAP…) But I think it’s a better choice considering it will be much easier to integrate all the other systems – activeCollab, Jabber, internal HR systems etc with MySQL.

It’s been a productive day! Looking forward to another exciting day tomorrow, when we move all the servers from LDAP to MySQL, and take the Asterisk setup forward!

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25 Comments

  1. Hi…

    I’m also deploying Asterisk via TrixBox. I’m having lots of problems with noise but not with echo via calls to pstn…. Can I ask, how did you overcome the noise problem with your implementation?

    best wishes
    Phil

  2. There is no noise when we use the Digium card. There is also no noise when we use the XLite client with noise reduction on. So the software can really remove the noise to a great extent. When I use SJPhone, and if the other person is on XLite, I can hear his voice very well, but he hears a lot of noise in my voice.

  3. Hello Nirav,

    On BSNL Pri line the TE110P card works fine with Asterisk.

    But, we are not able to configure the TE110p with tata indcom e1 pri line in our Mumbai office. This link is through MUX and not through modem. We are using straight connection rathar than cross cable.

    Could you let us know the content of /etc/zaptel.conf and /etc/asterisk/zapata.conf file?

    Regards

    pss

  4. Dear pss,
    I am from chennai using TE110P card with BSNL PRI Line. Would you let me know the configuration. My problem is Zttool states Yellow alarm.

  5. Hi Karthikeyan,

    When zttool shows yellow alarm, it means that the card is not getting proper signals from the ISDN line. Are you sure the cable is crimped correctly?

    In our experience, at all times, the yellow light / alarm is due to wrong crimping of the cable or a loose cable.

    :Nirav

  6. Hi!
    I have TE110P Digiun card,How do i configure this this inb trixbox for Reliance PRI line.Can some one explain in detail,I connected the cable and the LED’s are not glowing.Is it straight cable or crossover cable.

    Janardhana Reddy S

  7. Hi Janardhana,

    Most probably it is not a straight cable. Not exactly a crossover either.

    Mitul has posted an entry about how we did it for our Tata line. I believe that should work for you.

    If it does, please let me know. If it does not, try out some combination and see what works!

    Thanks.

  8. Hi! Nirav
    I tried ans still the LED’s are not glowing.
    Can you pleae let me know what should i do in zatel.conf.

    When i reboot the system it is finding new hardware and asking for IP address,shoul i give IP adress fro TE110P card?.

  9. Reading some of the posts in asterisk.pbx.in I realize that using VoIP gateway in India is illegal. An asterisk box connected with PRI essentially acts as a VoIP gateway…so does this mean that we can not use asterisk connected with PRI from any Indian telco? I’m not from telecom domain and all these terms have left me confused….can any one please help me understand these issues?

  10. Hi Ethan,

    There are some details involved. If you are using Asterisk for your EPABX and terminating the PRI line on it, it’s not a problem. If you route calls for others that would be a problem. The policies are quite liberal now, and VSNL even mentioned that if we take their leased line, we can even do international VoIP traffic on that legally!

    :Nirav

  11. Thanks Nirav…that saved me from a disaster! Also to update, we were able to connect with MTNL PRI using the cable that Mitul mentioned.

  12. Hi Ethan,

    Great. Is the cable crimping the same as the one for Tata? Would be great if you can post it up, so that others benefit.

    If you faced any other difficulties, you can mention the solution to them as well.

    :Nirav

  13. Hi,

    I am using Trixbox to set up an IP PBX for my office. We are currently using an Audio Codes MPX-108 FXO Gateway to connect to the PSTN.

    This gateway works fine with Airtel but when I connect to my Indicom FWT units, they take very long time to accept incoming calls and when I try to call out, I always get the message “The number you have dialed is incorrect” from the Indicom end.

    Would like to know if all providers in India use the same type of Dialing Methods or whether I need to make any changes from one provider to the other.

    Appreciate any inputs.

    Krish

  14. Hi All,

    My name is Raju I am from Chennai, i am very new to asterisk, i would like to implement asterisk, Please give me a guideline and also tell me where can i get the Hardware?

    Regards,
    Raju

  15. Hi,

    can anyone let me know if he is using reliance pri with asterisk.

    Please…wait to hear from you..need help

  16. how can i do to install a te110p with trixbox? i am novice ..and i think it s very interesting..
    regards!!!

  17. Airtel X100P configuration problems
    I am trying to configure X100P with my Airtel connection.
    I am running
    Zaptel 1.2.18
    Asterisk 1.2.20
    Libpri 1.2.4(i think)
    My zaptel.conf looks like

    fxsks=1
    loadzone = uk
    defaultzone = uk

    Zapata.conf looks like
    [channels]
    signalling=fxs_ks
    context=incoming
    channel=>1 ;X100P

    extentions.conf looks like
    [incoming]
    Exten => s,1,Wait(2)
    Exten => s,2,Record(/tmp/asterisk-recording:gsm)
    Exten => s,3,Wait(2)
    Exten => s,4,Playback (/tmp/asterisk-recording)
    Exten => s,5,wait(2)
    Exten => s,6,Hangup

    Now the problem is that on when I call to Asterisk, I always get busy
    tone from remote phone, what could be the problem? If I detach the line from the card I get ringing tone of course

    Asterisk Debug info:-
    cat /proc/zaptel/*
    results in
    Span 1: WCFXO/0 “Wildcard X100P Board 1”

    1 WCFXO/0/0 FXSKS (In use)

    *CLI> pri show span 1
    No PRI running on span 1

  18. hi, did you solve fxo busy tone issue? i have exactly the same proble as you.

    I am doing test at home.
    I use my mobile phone call my fixed line number.

    (1)If fixed line not connect to anything phone or fxo, i can hear the ring ton.
    (2)If i connect the fixed line to x100p, even x100p is not plug into PCI slot, i hear busy tone and the operator told me the line is engaged

    Who can help?

  19. Hi!
    Your noise issues are related to your codecs. I had the very same problem. My provider is BV and I was making and receiving calls ok. But, people would complain a lot when I would call them. I took a look at my codecs and I only had a couple enable. I did a google search to find out what codecs to enable and I found several. All I did was to go to my {general} section of my sip.conf and pasted the codecs; and now it works like a charm!

    Good luck to everybody!!!!

  20. hi friends,
    really i am new to asterisk server…
    from googling i installed a working asterisk server with x-lite client on our LAN
    I want to connect this to our bsnl land line and make and receive calls
    Also how can i achievw free call anywhere using asterisk
    kindly consider me as a newbie and tell me the easier steps…

    Thanks in advance

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