Re: [asterisk-users] Dahdi issue on sangoma A200

2010-09-02 Thread asteriskguru asteriskguru
Hi max,

Please read India-CID.txt file in asterisk documentation directory.

Thanks,
Ashik

On Wed, Aug 11, 2010 at 10:35 AM, Max Alex  wrote:

> 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 <
> beaasteriskg...@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  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]  
>>> 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
>>>
>>>
>>> --
>>> _
>>> -- 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
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>> --
>> _
>> -- 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Dahdi issue on sangoma A200

2010-08-10 Thread 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 <
beaasteriskg...@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  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]  
>> 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
>>
>>
>> --
>> _
>> -- 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
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- 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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Re: [asterisk-users] Dahdi issue on sangoma A200

2010-08-09 Thread asteriskguru asteriskguru
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  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]  
> 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
>
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
-- 
_
-- 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

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users