Displaying 20 results from an estimated 8000 matches similar to: "asterisk SIP MESSAGE method support"
2011 Apr 06
3
BRI Configuration help me
Sir,
i am using goautodial server , bri card is showing ok but when i try to call
that showing below ,
This configuration is in doing in dubai , so kindly help me how can connet
the call from this ,
what is my mistake is in this
:::chan-dahdi.conf
[channels]
#include
dahdi-channels.conf
language=en
context=default
usecallerid=yes
hidecallerid=yes
callwaiting=yes
usecallingpres=yes
2011 Jul 14
9
Extension wise dialplan
Hi all,
I have n no. of extensions in my dialer. from 456 to 556 extensions. I was
created 2 other extensions 667 and 668 I need to allow only STD calls to
go from this extensions.
These all extensions are same context . I need to define the STD dialplan
for only this 2 extensions. how I can ?
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI |
2011 Jun 01
10
busy hangup HDLC Bad FCS (8) on Primary D-channel
Hi all,
After running fine for a few months now asterisk seems to hang
frequently , still functioning but the DAHDI channels seem busy (users
report a busy signal when calling or being called)
A reboot will allow it to run for another day or maybe 2 or 3 till the
problem occurs again.
running stock Asterisk 1.6.2.9-2+squeeze2 on Debian with stock kernel
2.6.32-5-686
i get the following
2011 May 09
3
OUTBOUND CALLER ID
Hi,
THIS IS IN DUBAI.
I am having PRI line with 100 DID's (00-99) and when we call to any landline
or mobile number then it shows us our board number or pilot number (i.e
4663000 means 00).. As i give all the extensions a particular DID, so people
from outside world can call them. The problem is the CALLERID ... When we
call from any of other extension PSTN line carries out our pilot number
2011 Jun 07
3
Different callerid for different extensions
Hi,
I have small confusion in my configuration which is I had some DID's like
044578900-04457999. I was configured dial plan below mention.
exten => _0XXXXXXXXX,1,NoOp(Int exten:${CALLERID(num)})
exten => _0XXXXXXXXX,2,Set(outgoing_ident=0445789${CALLERID(num):-2})
exten => _0XXXXXXXXX,3,NoOp(Ext ident:${outgoing_ident})
exten =>
2011 Apr 11
1
Require dialplan
Hi ,
In vicidial dialer
I need small Dialplan require. when i call from hardphone , in that has 1to9
no.s i want define the dipositions like when i press the 1 it will goes
NotIntrest, press 2 for NotAvailable.
How can i configure for this.
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75
2011 Jun 13
3
asterisk queue 'ringall' stratagy
Hi List,
I have faced a problem in asterisk queue implementation.
I configured a queue with 'ringall' strategy and 'ringinuse=yes' in queues.conf. If three calls come to this queue in parallel, the logged in queue agent used to get only one call (may be the first one), not all the calls waiting in the queue at a time. Once the agent answers the call the next call is displayed.
I
2011 May 11
2
no audio with SIP:INFO in meetme
Hello List,
Asterisk is blocking audio if 'F' flag is enabled in meetme with DTMF mode enabled as INFO for SIP channel.
If it is a bug in asterisk or something need to be enabled in sip.conf for the same.
Dialplan looks like
Exten => 100,1,MeetMe(100,dmF)
Sip.conf
dtmfmode=info
Regards,
Rajib
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery
2011 Apr 20
2
No voice in MeetMe for SIP with
Thanks a lot Tony and Dhaval for your much appreciable suggestions.
Regards,
Rajib
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
www.siemens.com
Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com
Date: Wed, 20 Apr 2011 13:55:25 +0530
From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>
2011 Apr 19
3
No voice in MeetMe for SIP with AGI_BACKGROUND
Hello List,
I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND.
http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.
2011 Apr 19
1
ConfBridge and AGI
Hello List,
Is it possible to run an AGI script in backgroung for all the associated SIP channels in ConfBridge Application? If yes how?
This can be done using 'b' parameter in MeetMe for non SIP channels.
Regards,
Rajib
Rajib Deka
SIEMENS Ltd.
Robert V Chandran Tower, First Floor, West Wing,
#149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA.
2011 Jun 16
2
Inbound call not dialing exten
Hi all,
I have 100 DID's which is 4578900 to 4578999 , and i have 5001 to 5099
extensions. when incomming call come to this DID no. (4578901) that time
5001 extestinsion should ring.
below my dial plan is not getting any result , inthat has any mistakes.
please help.
exten => _45789XX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _45789XX,1,Set(Dest=2{EXTEN:-2})
exten =>
2011 Apr 01
2
BRI detection
Hi,
I need to configure BRI 4span card in dubai in vicidialnow for dialer
perpose. in that i have small confusion which is NT an TE mode . that was i
am setting perfectly but dubai telco what they are use for this i dont know
which parameters are use for that . please help me.
--
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD |
2013 Sep 03
3
Asterisk crash
Hello List,
In our lab asterisk has crashed due to some unknown reason and it has been restarted by safe_asterisk service. But before crash we can see lots of below log entry (asterisk version 1.8.9.3).
Sep 3 07:55:21] WARNING[16287] res_rtp_asterisk.c: RTP Transmission error of packet to [2002:c117:a683::c117:a683]:20940: Address family not supported by protocol
chan_sip.c: Purely numeric
2011 May 04
2
asterisk HA for queue calls
Hello List,
We are running two asterisk machines in virtual IP as primary and secondary server.
Initially virtual IP will be active in primary server; during the failure of primary secondary will get the virtual IP.
Is there any way to retrieve pending queue calls from primary to secondary, in case primary fails?
Does asterisk provide any interface to do it or we have to write some application
2011 Jun 16
1
#include filename
Hi,
I am using asterisk1.2
In this, my dialplan is going large , so i need to configure this small
pieces for this, i did in my extensions.conf
when I dial the 123 its not going , means that file is not reading. is there
any parameters to add any where ? please tell me
this #include is not working ...
extensions.conf
[general]
[global]
trunk=zap/g0
#include exten-internal.conf
[default]
exten
2011 May 25
2
asterisk hint SIP presence
Hello List,
Asterisk CLI command "core show hints" gives the list of hint extension configured and its presence status.
In command output there is a field called "watchers" and it contains a numeric value of number of subscriptions' registered for that particular extension.
So, is there any CLI command to check who the watchers for an extension are?
Regards,
Rajib
Rajib
2011 Jun 08
1
CallerID issue
Hi List,
I am making outgoing call from asterisk to GSM network with the help of VoIP
trunk(SIP trunk) then I am not geting any caller ID at destination end. Is
this the asterisk issue or VoIP trunk issue?
Is this is due to asterisk then how we solve it? I already user
Set(CALLERID(num)=XXXXXXXXXXX) in dialplan.
-----
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
2011 May 30
3
please help
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten => _0678922645.,1,Set(CALLERID(number)=520460587)
exten => _0678922645
.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => _0678922645
.,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))
exten => _067892264*5*,2,Hangup()
i can not call my
2014 Aug 01
1
Connecting Asterisk and BT Versatility PBX via NT BRI port
Hi All,
I've a BT Versatility PBX that I want to connect to my asterisk 11.9.0 box via BRI port in NT
mode but actually I wasn't able to get it working. I've another standalone PBX, it's a
Panasonic model, which works fine connected to the same port. The BT connected to a UK
BT BRI line works quite fine after roughly a minute once connected. The PBX can be
seen at