[asterisk-users] Dahdi issue on sangoma A200

[Max Alex]

Hi,
Thanks for this information, but it is not working for both the issues,
I have tried with the configuration with cidsignalling, cidstart etc..
Can any one provide more help for this.

Thanks,
Max Alex
Voip Developer



On Mon, Aug 9, 2010 at 5:31 PM, asteriskguru asteriskguru <
beaasteriskguru at gmail.com> wrote:

> Hi max,
> Have look on my blog regarding this.
>
>
> http://ashikalim.blogspot.com/2010/08/config-asterisk-india-pstn-lines_09.html
>
> Thanks,
> Ashik
>
> On Sat, Aug 7, 2010 at 11:15 AM, Max Alex <max.asterisk at gmail.com> wrote:
>
>> Hi All,
>> I have Sangoma A200 Card installed on my system,
>> I have centos 5.5 with 64 bit,
>> Here are the description for asterisk and dahdi.
>> Asterisk 1.6..2.9
>> Dahdi: 2.3.0.1
>> I have two issues with dahdi
>> 1) I am not getting full callerid on my phones from sangoma card to
>> asterisk users. if i am connecting analog phone directly then i am getting
>> callerid properly.
>> I am in india and using Airtel Connection, I have set variables in
>> chan_dahdi.conf as well for callerid but the not getting full digits in
>> callerid,
>> it is coming with 8 digits only.
>> 2) Another issue is when I am hanging up the phone from inbound or
>> outbound from the dahdi channel, it takes 5-6 seconds to dropping the call.
>>
>> Here are the confguration file for chan_dahdi.conf
>> -------------------------------------
>> ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
>> ;autogenrated on 2010-07-30
>> ;Dahdi Channels Configurations
>> ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak
>>
>> [trunkgroups]
>>
>> [channels]
>> context=default
>> usecallerid=yes
>> callerid=asreceived
>> hanguponpolarityswitch=yes
>> answeronpolarityswitch=yes
>> ;cidstart=ring
>> cidstart=polarity_IN
>> ;cidsignalling=dtmf
>> cidsignalling=dtmf
>> hidecallerid=no
>> callwaiting=yes
>> usecallingpres=yes
>> callwaitingcallerid=yes
>> threewaycalling=yes
>> transfer=yes
>> canpark=yes
>> cancallforward=yes
>> callreturn=yes
>> echocancel=yes
>> echocancelwhenbridged=yes
>> relaxdtmf=yes
>> rxgain=0.0
>> txgain=0.0
>> group=1
>> callgroup=1
>> pickupgroup=1
>> immediate=no
>> useincomingcalleridondahditransfer=yes
>> ;callerid=asreceived
>>
>> ;Sangoma AFT-A200 [slot:4 bus:2 span:1]  <wanpipe1>
>> context=from-internal
>> group=1
>> echocancel=yes
>> callerid=asreceived
>> signalling = fxo_ks
>> channel => 1
>>
>> context=from-internal
>> group=1
>> echocancel=yes
>> callerid=asreceived
>> signalling = fxo_ks
>> channel => 2
>>
>> context=from-zaptel
>> group=0
>> echocancel=yes
>> callerid=asreceived
>> signalling = fxs_ks
>> channel => 3
>>
>> context=from-zaptel
>> group=0
>> echocancel=yes
>> callerid=asreceived
>> signalling = fxs_ks
>> channel => 4
>> -------------------------------
>> Please hemp me for this issues.
>>
>> Thanks,
>> Max Alex
>> Voip Developer
>>
>>
>> --
>> _____________________________________________________________________
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>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
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