Hi Rilawich,
I worked recently on it and that is why can give you the idea how i achived it.
You can write an PHP script to get the number and name of the
customer.You can phpself to the script.Then you can use an API script
to use that number to orignate the call.The channel will be used to
call
On Tuesday 06 January 2009 00:11:22 Rilawich Ango wrote:
> Hi all,
>
> I want to build a web page for user to input a phone number. Then,
> the number will input to asterisk and it will makes call. At that
> moment, asterisk will make another call to a internal ext. Finally
> asterisk will bri
Check out the Manager interface.
Rilawich Ango wrote:
> Hi all,
>
> I want to build a web page for user to input a phone number. Then,
> the number will input to asterisk and it will makes call. At that
> moment, asterisk will make another call to a internal ext. Finally
> asterisk will bri
Hi all,
I want to build a web page for user to input a phone number. Then,
the number will input to asterisk and it will makes call. At that
moment, asterisk will make another call to a internal ext. Finally
asterisk will bridge 2 calls together for conversion.
Does asterisk can do it? How?
Murf,
Thanks for the update. I look forward to seeing this one resolved. This
is just the issue that I'm facing. Looks like there's a patch already
posted on the bug. I'll wait for the bug to be closed or pushed to
release. Thanks again.
Robert
Steve Murphy wrote:
> On Mon, 2009-01-05 at 12:
Hello list,
Has anyone on list had experience with getting the 7914 module working
with Asterisk, probably using the chan-sccp drivers?
I have 7912 and 7940 phones working OK, I need something for
reception, and I was thinking a 7960 with two of these modules would
handle it, but need to know tha
I tried that before, but I just tried it again. Unfortunately, the same
thing:
No user '5551236049' in SIP users list
Found peer 'ACME' for '5551236049' from 172.16.10.40:5060
[ACME]
host=172.16.10.40
username=username
secret=password
type=friend
Frank
From: asterisk-users-boun...@li
On Mon, Jan 5, 2009 at 10:12 PM, David fire wrote:
>
> if you don't know any parser maybe you can send me a link or a pdf whit info
> on how to parse the log.
>
...check queuelog.txt under the doc/ directory on the asterisk source
distribution (apparently, under 1.6 it is queuelog.tex... no m
Try it by IP address instead of hostname as reverse DNS may not be
resolving. e.g. host=123.123.123.123
On Tue, Jan 6, 2009 at 2:25 PM, Frank Bulk wrote:
> This is what I have in my configuration now:
>
> [ACME]
> host=sip.acme.com
> username=username
> secret=password
> type=friend
>
> I've don
On Mon, Jan 5, 2009 at 8:20 AM, Nick Wolf wrote:
>
> besides this, I paste my zaptel.conf :
>
> span=1,1,6,ccs,hdb3
> span=2,1,6,ccs,hdb3
> bchan=1-15,17-31
> dchan=16
> bchan=32-46,48-62
> dchan=47
> loadzone=fr
> defaultzone=fr
>
> when I put 6 in LINE BUILD OUT (1,1,6,ccs,hdb3) value I got bett
This is what I have in my configuration now:
[ACME]
host=sip.acme.com
username=username
secret=password
type=friend
I've done a SIP debug before, but I've done it again with the above
configuration:
No user '5551236049' in SIP users list
Found peer 'ACME' for '5551236049' from 172
Well, it's not "acme.com", but another domain. That information about the
encoding process of the nonce is helpful to know.
Do I need to specify the context to be "sip.acme.com"? Where is that
"acme.com" specified in the trunk configuration?
Frank
-Original Message-
From: Alex Balashov
Frank Bulk - iName.com wrote:
>The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not
>work unless I add "insecure=very" to my "Outgoing settings", but I don't
>want to do that. I do want to authenticate. Outgoing (Asterisk PBX to
>Class 5 switch) calls do authenticate and wo
On Mon, 2009-01-05 at 12:27 -0700, Robert Broyles wrote:
> On 12/17/08 I updated to 1.4.22 from 1.4.21...
>
> Now the CDR data isn't recording calls where the caller hung up while
> waiting on the Queue.
>
> Sample CDR data BEFORE the upgrade:
>
> "2008-10-30 12:46:47";"\"John\"
> <0006741103>
Is sip.acme.com actually the domain you want to use?
Keep in mind the domain is part of the digest authentication process and
is a factor in the encoding of the nonce.
Frank Bulk - iName.com wrote:
> The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not
> work unless I add
On Monday 05 January 2009 16:52:28 Danny Nicholas wrote:
> Yes, but if you do, you will lose it in a future upgrade (if that matters
> to you).
No, he won't. Our current policy is that while we may deprecate
functionality, we will never again remove it (unless the deprecated
functionality somehow
The incoming (Class 5 switch to Asterisk PBX) side of a SIP trunk does not
work unless I add "insecure=very" to my "Outgoing settings", but I don't
want to do that. I do want to authenticate. Outgoing (Asterisk PBX to
Class 5 switch) calls do authenticate and work.
The Nortel CS 1500 I'm using a
If you don't want to use the AEL, but want an easy way to have agents
login and out, check out this quick tutorial:
http://hostseries.com/agentcallbacklogin-alternative/
Ariel Dorfman wrote:
> i have done some research, but there says that i can use a function called
> AgentCallbackLogin, bu
Yes, but if you do, you will lose it in a future upgrade (if that matters to
you).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pedram M
Sent: Monday, January 05, 2009 4:41 PM
To: Asterisk Users Mailing List - Non-Commerc
Ariel,
You can still use it eventhough it is deprecated.
Read up on the following:
http://www.voip-info.org/wiki-Asterisk+call+queues
http://www.voip-info.org/wiki-Asterisk+agents
http://www.voip-info.org/wiki-Asterisk+cmd+AgentCallbackLogin
http://www.orderlyq.com/asteriskqueues.html
Regards,
> As the subject says, I need to implement on my call center the Agent
> functionality, son the agents could logon and logoff to the queue
> How can I do this configuration? Or where can I read some info about
it
Here is a few links I used when I developed mine.
http://www.voip-info.org/wiki/vie
i have done some research, but there says that i can use a function called
AgentCallbackLogin, but it is deprecated in my system and i cant use it
regards
>Ariel Dorfman a écrit :
>>
>> Hi all
>>
>> This is my first post.
>>
>> As the subject says, I need to implement on my call center the Agent
hi
i am integrating asterisk whit an open source CRM or maybe i am integrating
an open source crm whit asterisk any way
the CRM is vtiger, (www.vtiger.com)
i need an opensource queue log parser to put inside the crm agent statics.
i need any parser i can port it to php or to java
if you don't know
Benoit wrote:
Brent Davidson a écrit :
Another question along these lines... If I set a Global called
"TRUNK" in the globals section and later assign do a TRUNK=whatever it
appears that a local variable called TRUNK is created instead of using
the global. You must explicitly use the Set(GLO
Brent Davidson a écrit :
> Watkins, Bradley wrote:
>> ...
>>> Well, before I file a bug I have another question... In AEL,
>>>
>> what is the correct syntax? Do all variable references still need to be
>> wrapped in ${} or not? If they do, then the documentation on
>> voip-info.org need
Ariel Dorfman a écrit :
>
> Hi all
>
> This is my first post.
>
> As the subject says, I need to implement on my call center the Agent
> functionality, son the agents could logon and logoff to the queue
>
> How can I do this configuration? Or where can I read some info about it
>
> Regards
>
> Ari
Watkins, Bradley wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brent
Davidson
Sent: Wednesday, December 31, 2008 1:03 PM
To: m...@digium.com; Asterisk Users Mai
Hi all
This is my first post.
As the subject says, I need to implement on my call center the Agent
functionality, son the agents could logon and logoff to the queue
How can I do this configuration? Or where can I read some info about it
Regards
Ariel
_
On 12/17/08 I updated to 1.4.22 from 1.4.21...
Now the CDR data isn't recording calls where the caller hung up while
waiting on the Queue.
Sample CDR data BEFORE the upgrade:
"2008-10-30 12:46:47";"\"John\"
<0006741103>";"0006741103";"11621708182";"incoming";"SIP/carrier-3";"SIP/120-09232ae0";
On Monday 05 January 2009 11:40:07 JR Richardson wrote:
> Hi All,
>
> I have some Asterisk 1.2 servers using the cdr_mysql addon (1.2.3)
> spitting cdr's over to a MySQL database on another server. All is
> working well except for a strange problem I ran into this morning.
> During some cdr databa
Hi All,
I have some Asterisk 1.2 servers using the cdr_mysql addon (1.2.3)
spitting cdr's over to a MySQL database on another server. All is
working well except for a strange problem I ran into this morning.
During some cdr database maintenance, the cdr table was locked for a
few minutes, during
Olivier Fauchon a écrit :
> Hi.
>
> When I call my RNIS numbers (with a mobile phone for example), I can
> see 2 incoming calls on the IPBX, which should not happend.
>
> I'm not sure if it's a problem with the telco France Telecom and their
> ISDN setup, or if it's a problem
> with the MISDN dr
Hi,
My setup is SIP Call-->Asterisk-->VSP1 or VSP2 or VSP3
I'm experiencing an interconnect issue with one of the VSP's that seems to
have to do with Asterisk not having any VAD control. The error is:
NOTICE[10989]: frame.c:203 __ast_smoother_feed: Dropping extra frame of
G.729 since we already
I have a sata harddrive, do you think changing it to ata one will
solve the problem ? any one has tried this solution ?
___
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To UNSUBSCRIBE or update option
Hi.
When I call my RNIS numbers (with a mobile phone for example), I can
see 2 incoming calls on the IPBX, which should not happend.
I'm not sure if it's a problem with the telco France Telecom and their
ISDN setup, or if it's a problem
with the MISDN driver on the IPBX itself.
I'm stuck ...
and here is lspci output:
lspci
00:00.0 Host bridge: Intel Corporation Unknown device 29f0 (rev 01)
00:01.0 PCI bridge: Intel Corporation Unknown device 29f1 (rev 01)
00:1c.0 PCI bridge: Intel Corporation Unknown device 2940 (rev 02)
00:1c.4 PCI bridge: Intel Corporation Unknown device 2948 (rev 0
Hi and Thanks for your reply,
I am sticking to 1.2.30 because this is a Vicidial server & they
recommend this version.
With 10 to 15 concurrent calls load on the server is 0.2 sometimes
less, (thats why I was told that I don't need to recompile asterisk
since I have no cpu overloading)
but here
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