Hi All,
We made a VOIP application for PDA's (PALM OS) and we are using both SER
and Asterisk. SER is SIP proxy and it routes all the calls to Asterisk. On SER
we have RTPProxy also. My problem is that I am getting a weird noise or
disturbance for all the calls at an approximate time
Hi All,
We are making a VOIP application for Mobiles (PDA's) and we are using
Asterisk
for it. We have a setup consisting of both SER and Asterisk. SER acts as a SIP
router and routes everything to Asterisk. We also have rtpproxy for SER. Our
packet delivery from clients (Mobiles, PDA's)
Hi Benchev,
Thanks a lot for your replies.
I understood that without mentioning context names in Extensions.conf
we cannot
configure contexts in Asterisk Realtime.
Thanks and Regards,
Manoj.
Quoting Benchev [EMAIL PROTECTED]:
I need many contexts because I have around 1000 DID's each
Hi All,
Thanks for your replies.
I need many contexts because I have around 1000 DID's each with 5-10
Extensions.
These DID numbers are changed or added very frequently and whenever there is a
change I have to change Extensions.conf manually. So please tell me how can I
do this dynamically
Hi All,
I will again tell what I am trying to do.
I have around 1000 DID's and I have to setup context for each of it's
extension
and I want to do that dynamically and I do not want to change extensions.conf
all the time manually whenever I want to add new context instead I will do it
in
Hi Benchev,
Thanks for the reply.
My current setup is exactly similar to which you have suggested. My DID
numbers
are added or changed very frequently and all the time I have to change some
config file manually and should reload Asterisk or atleast call Extensions
reload. I do want these
Hi All,
I was able to install Asterisk and Asterisk-addons and use them
successfully.
But I have a problem now, I have many contexts and it looks like Asterisk is
unable to find the context given directly in Mysql DB unless I specify it in
Extensions.conf to switch it to RealTime. If I
Hi,
Thanks for your replies.
I am going to have many DID's and I have to provide each of them this feature.
So I cannot solve this problem with a dedicated DID having G711. Is
there a way
to change codecs in the middle of the call? Please tell me what else can I do
here?
Quoting Darrick
Hi All,
I was able to insert some extensions in Mysql DB and use them successfully. In
Mysql extensions table the priority column is of type tinyint and when I give
's' value for it, it is not accepting that value as it takes only tinyints.
Please tell how can I make that column accept values
Hi All,
I want to configure fax with Asterisk and I found that we can do this reliably
using G711 codec only. Currently my provider is supporting G729 and G711.
During the call initiation the call starts with G729 (1'st priority) and
somehow if the receiver is unable to receive call then we are
Hi Group,
I was able to install Asterisk and its addons successfully. Now I want to
eliminate sip.conf and extensions.conf and use everything from Mysql DB, Is
this possible? I have seen this page
http://www.voip-info.org/wiki/index.php?page=Asterisk%20extensions%20from%20mysql
and learnt that
Hi Group,
Please read my previous message below, I want to configure Asterisk with Mysql
and make Asterisk dynamic so that Asterisk will read everything from Mysql and
we can make changes to mysql data directly. Please tell how can we do this and
point me to related documentation.
Thanks for
Hi All,
I installed Asterisk recently and it was working from 2 weeks without a problem
until today. Today it started showing strange error
Feb 28 03:14:08 WARNING[31430]: chan_sip.c:4826 check_auth: Stale nonce received
from 'sip:[EMAIL PROTECTED]'
Whatever number I call it displays this,
Hi All,
I was able to install Asterisk and make outgoing calls. Recently I purchased two
DID's and I am facing a problem configuring them to my Asterisk, I hope with
the help I get from this list I will be able to configure successfully. Mu
errors are
Feb 28 08:31:58 NOTICE[19133]: pbx.c:1331
Hi All,
I am a newbie to Asterisk and I was able to install Asterisk and call out.
Recently I purchased two DID's, can someone please tell me or point to some
links showing how to configure these DID's for SIP based softphones like
Express talk?
Thanks,
Manoj.
Hi,
I have a configuration which is working fine and for that SER is used
ONLY as a
proxy/registrar, and all calls are routed to Asterisk so that Asterisk places
the calls to the PSTN?
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Hi,
We have a task to reduce voice call bandwidth. IP+UDP+RTP are using 40 bytes per
packet and for voice GSM FR 33 bytes. We are trying to reduce this bandwidth
accommodating multiple GSM frames in one packet. If we want to use per packet
10 GSM frames how to do this using asterisk? Assume the
Hi All,
I configured Asterisk and it is working successfully with Express Talk. Now I am
trying to work with some other client which supports only GSM and now Asterisk
never worked and tried to make a call out. In sip.conf I disallowed all and
allowed only GSM also. I also heard that Asterisk
Hi All,
I tried with different configurations and referred many articles to configure
Asterisk with a Vonage account I have but all my attempts failed. I am a newbie
and hope this mailing list will help fixing my problem and configure Asterisk.
The error I get after I make a call to outside
Hi All,
I have installed Asterisk and able to create Users and get them connected to
Asterisk after authentication. My question is how can I make calls to different
DIAX clients through my Asterisk server. I also have vonage softphone account,
using that I tried calling 18882255322
--
Hi All,
I installed Asterisk and trying to configure Vonage with it. After getting
authenticated when I try to call to a number I get the following errors
First I get
Sip read:
SIP/2.0 407 Proxy Authentication Required
CSeq: 104 INVITE
Proxy-Authenticate: Digest realm=216.115.20.41,
Hi All,
I am a newbie to VOIP and after some problems I was able to install Asterisk. If
I start Asterisk I could find Asterisk Ready at the end and I am thinking
that Asterisk is started successfully. Later after changing my Extensions.conf
and ser.conf nothing works, I could still see the
Hi All,
After few problems I have installed Asterisk and changed my iax.conf. I have
defined a user in iax.conf and when I try to connect that user from DIAX phone
I get the following error
Jan 17 23:48:16 NOTICE[16448]: chan_iax2.c:3910 register_verify: No registration
for peer 'manoj' (from
Hi All,
I am a newbie and trying to install Asterisk from instructions given in
http://www.voip-info.org/tiki-index.php?page=Asterisk+RPM. We have Centos 3.3 so
I downloaded rpm's from
ftp://ftp.linuxsys.com/pub/LSE/packages/CentOS-3.4/asterisk-1.0.9/ and tried
installing one by one but I get the
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