Displaying 14 results from an estimated 14 matches for "faheem2084".
2016 Sep 15
2
Asterisk 13 externip
On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com>
wrote:
>
>
> On Wednesday, 14 September 2016, Madushan Geethanga <
> mgliyanage.rc at gmail.com> wrote:
>
>> Hi,
>>
>> What is the equal option for externip in asterisk 13 with pjsip. I have
>> tried
>>
>> external_media_add...
2016 Sep 15
2
Asterisk 13 externip
...to respond back to the address in the From header
instead of the Contact header?
>
> Best Regards,
> Madushan
>
> On Thu, Sep 15, 2016 at 7:41 PM, George Joseph <gjoseph at digium.com> wrote:
>
>>
>>
>> On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com>
>> wrote:
>>
>>>
>>>
>>> On Wednesday, 14 September 2016, Madushan Geethanga <
>>> mgliyanage.rc at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> What is the equal option for externip in a...
2016 Sep 16
3
Asterisk 13 externip
...t;>>
>>> Best Regards,
>>> Madushan
>>>
>>> On Thu, Sep 15, 2016 at 7:41 PM, George Joseph <gjoseph at digium.com>
>>> wrote:
>>>
>>>>
>>>>
>>>> On Thu, Sep 15, 2016 at 7:17 AM, Faheem Muhammad <faheem2084 at gmail.com>
>>>> wrote:
>>>>
>>>>>
>>>>>
>>>>> On Wednesday, 14 September 2016, Madushan Geethanga <
>>>>> mgliyanage.rc at gmail.com> wrote:
>>>>>
>>>>>> Hi,
>>>...
2016 Sep 14
2
Asterisk 13 externip
Hi,
What is the equal option for externip in asterisk 13 with pjsip. I have
tried
external_media_address=XX.XX.XX.XX
external_signaling_address=XX.XX.XX.XX
but asterisk 13 writes local ip to the from header. any suggestions?
Best Regards,
Madushan
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2013 May 16
1
Call Transfer question
Hi,
is possible that two sip extensions: user-1 and user-2 are connected and I
want to transfer the call from user-1 to a third user "user-3".
I know it is possible through feature keys mapping in features.conf, but I
want to do this through AMI or Asterisk CLI Commands?
Please suggest if possible?
Thank you!
Muhammad Faheem
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2016 May 18
2
variable to get waittime of caller exiting queue
Hi all
Is there anyway i could get in the dialplan the amount of time a caller
waited in the queue before exiting?
Thanks
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2015 Sep 17
2
I want to store cdr into database
I have asterisk 13.5 configured with a simple dial plan, 3 SIP clients two
Laptops and smartphone with softphones installed. Now I am trying to store
cdr into a database but not able to make a connection of ODBC drivers to
MySQL is there an option or anything. Thanks in advance
My configuration::
*sip.conf*
[general]
trasport=udp ;Data format | sample commennt
[template01](!)
type=friend
2015 Sep 14
2
AgentLogin() on the multiple servers?
Hello,
Let say all the SIP devices will be registered on the proxy like kamailio.
Agent is a member of Support and Billings Queues on the asterisk servers.
Support queue on "Server A" and Billings Queue on "Server B" for example.
This will be done via RealTime Queue.
I want Agent to dial 1234 on a sip device and it will prompt to enter a pin
number to Login via
2016 May 03
2
Is MixMonitor command is blocking ?
Hello,
I try to find informations concerning Mixmonitor command, but ... without
success.
MixMonitor command take at last parameter "command". This command can be a
shell script.
When record is over, and this command executed, asterisk wait for a return
code or asterisk move to the next dialplan instruction ?
This command is a background task or use ressources in asterisk ?
For
2016 Jul 20
3
PJSIP_DIAL_CONTACTS issue
Hi,
I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS.
When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial
command breaks and the call control go to hangup block instead of next
priority. The error in CLI says "*Dial requires an argument
(technology/resource)*".
This error seems legit as there are no contacts for an offline endpoint.
The dialplan
2016 Jun 07
2
Want to detect sound
<!DOCTYPE html>
<html><head>
<meta charset="UTF-8">
</head><body><p>Hello everybody,<br><br>I manage not to detect one silence with record () when I make as follows:<br><br>Exten = > 0178900271, n, Record ($ ${ link_recorded_pseudos_clients } pseudo_ Client_Id} wav, 5,5) exten = > 0178900271, n, GotoIf ($ ["
2016 Jun 07
2
Delay after Answer
Well, I thought I had the problem solved. Ported everything over to
PJSip and build RDNS records for the phones and the server, but I am
still experiencing the problem on incoming calls.
**
On 6/7/2016 1:00 PM, Faheem Muhammad wrote:
> I've faced the same issue. The issue was related to DNS, the reverse
> lookup query failure caused the delay around(7-9 seconds). The purpose
>
2013 Dec 11
0
invalid From/Contact header values
Hi,
I'm observing wrong From/Contact header values. When I try to set
CallerID(num) it has no effect in the From and Contact Headers, and these
values are the same as the dialed number.
SIP Peers are defined using asterisk realtime. If I define the SIP Peers
using sip.conf then From/Contact header value are correct.
extentions.conf
[test]
exten=> 1000, 1,NoOp()
same=>
2013 May 11
1
AMI Originate issue
Hi,
I'm getting an issue while executing AMI Originate.
I'm getting "extension does not exists" on Originate's Response, and on the
other hand Asterisk CLI say "fwrite() returned error: Broken pipe"
Please suggest me what is wrong.
Muhammad Faheem
### my originate code block ...