Hello,
We are using Asterisk-1.0.3 version. Grandstream works
fine with this version. When trying to use a Sipura 2000 we the 488
error and also Answering with non-codec capability 0x1
(telephone-event) message.
also in the sip show peer for Sipura
Codecs : 0x0 (nothing)
Codec
Hello All,
We have VOIP running on Asterisk.
We are having problems with voicemail. As in when a customer is
having more than 1 phone numbers and he wants to check his mailbox
when he dials in to check his voicemail it should pick the primary
mailbox but it is picking up the w
Around 1 customers.
On Fri, 10 Dec 2004 17:24:56 +0100, Wilson Pickett
<[EMAIL PROTECTED]> wrote:
> How many customers, Sharon?
>
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Hello All,
I am receiving following error message while debugging Asterisk with
gdb.Help Appreciated
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 16384 (LWP 7710)]
0x4018f0f1 in strncpy () from /lib/libc.so.6
(gdb) bt
#0 0x4018f0f1 in strncpy () from /lib/libc.so.6
#
In extensions.conf
exten => _1NXXNXX,1,MySQLExtension(SIPE=${EXTEN:1}/${CALLERIDNUM})
exten => _1NXXNXX,2,GoToIf($[empty${SIPE} = empty]?6:3)
exten => _1NXXNXX,3,MySQLExtension(MATCH=${EXTEN:1}/${CALLERIDNUM})
exten => _1NXXNXX,4,GoToIf($[empty${MATCH} = empty]?5:20)
exten => _1NXX
Hello,
when SER redirects calls to asterisk can it be redirected to a
realtime peer in asterisk.i cld redirect the call to a static peer but
if someone can guide me through for realtime peer settings.
Thank you,
AA
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hello,
can someone help me with ser redirect to asterisk.
any help appreciated.
Thanks,
AA
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my setup is
client--registers--> ser-redirect--->client ---invite--> asterisk --> pstn
when this happens
i configured the ser.cfg with the rewriteuri and redirect logic and i
am seeing 300 redirect being passed to the client registerd to ser but
when it sends a invite to asteris
ing error
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to xxx.xxx.xx.xx : 5060 (NAT)
chan_sip.c:realtime_peer: Cannot Determine peer name ip=xx.xxx.xxx.xxx
Found no matching peer or user for 'xx.xx.xx.xx:5060"
its looking for same ip of the ser client to send back the reply.
Hello all,
I have setup my Asterisk and SER boxes.
implementing the ser.cfg and extensions.conf logic i am able to make
calls from asterisk to ser and vice versa. is it possible to make
simultaneous calls to a ser client from different asterisk clients
without getting a 486 busy from SER.
Thank
hello,
I am trying to pass MWI from Asterisk to SER.my user agents register
with Ser.i am not able to figure out how to do this.
i added the changes for mailbox in sip.conf for ser peer entry.
[ser]
type=friend
mailbox=XYZ
also changes in chan_sip.c for asterisk but not seeing the notify
mes
Hello,
In our company we are using Asterisk-cvs-Head with realtime. I am not
able to figure out a issue wherein the sip channel stays active
indefinitely.This happens when a call is in progress and the person
who called doesn't hangup normally but in between the conversation his
ATA gets unregiste
Steve,
I have been experiencing the same issue. I tried the qualify option as
well but didn't help.
Let me know if you find any solution.
-AG
On 8/23/05, Steve Edwards <[EMAIL PROTECTED]> wrote:
> I have Sipura 841's talking to a CVS-HEAD of August 4.
>
> If I disconnect the power to the Sipura,
Hello,
Is there a way to restrict realtime to not set callerid via
sippeers table even if there a column callerid in that table.something
like restrictcid in sip.conf
Thank you,
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Aster
I am trying on customizing the voicemail email settings per user ,the
settings being derived from the database.i was able to customize most
of it but the serveremail settings i'm not able to customize. while
deriving a string from databse which gives the from_email_ address and
assigning it to serv
I am using a A104 Sangoma card. We are runningasterisk cvs head on our
production box.After wanpipe configuration I am receiving the below
mentioned error.
pri show span looks good as below.
pri show span 1
Primary D-channel: 24
Status: Provisioned, In Alarm, Down, Active
Switchtype: National ISDN
number of cards.
Thanks,
On 10/25/05, Matt Florell <[EMAIL PROTECTED]> wrote:
> Have you tried Asterisk 1.2beta1? does it work under that release?
>
> We have been using an a104u with PRIs on 1.2b1 for about 6 weeks now
> with no problems.
>
> MATT---
>
> On 10/25/0
ure if that's an option, but
> when I compile, I use the option 3 and the pri only needs the channels and
> the d chan specified. So I never noticed if it said type 2.
>
> :)
> Paul
>
>
>
>
> -Original Message-
> From: Sharon [mailto:[EMAIL PROTE
Hello,
We are running asterisk cvs head on our servers. We are having
issues with the voicemail getting locked for some users when opposite
person tries to leave a voicemail. This happens randomly .Error
message seen on the server:
ast_lock_path: Failed to lock path
/var/spool/aster
TED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Sharon
> Sent: Tuesday, November 15, 2005 9:42 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] voicemail locking
>
> Hello,
> We are running asterisk cvs head on our serv
We are running the latest stable version of asterisk
Is there a way yet to load the realtime peers automatically from
database like the reload command does .
Everytime we make a change to the peer we have to manually load the peer
using
sip show peer abc load
Any suggestions appreciated,
Thank yo
I have my peers registered to SER.asterisk seems to be sending mwi for
the peers seen in the sip show peers CLI command. i have my ser server
registered with asterisk as a type=friend and all clients register to
ser.how do i get mwi to work for these clients registered to SER.
Thank you,
-AA
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-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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Hi there, I am looking around, is there anyone did any integration asterisk talk to / connect to MS SQL? Thanks-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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the DB through IVR. Any guidance?
Thanks. On 11/14/06, Vicky <[EMAIL PROTECTED]> wrote:
oops sorry i thought its my sql didnt notice it's MS SQL :D On 14/11/06, Tony Mountifield <
[EMAIL PROTECTED]
> wrote:
In article <[EMAIL PROTECTED]
>,Sharon Lim <
[EMAIL PROTECTED]&
D]> wrote:
In article <[EMAIL PROTECTED]>,
Sharon Lim < [EMAIL PROTECTED]> wrote:
> -=-=-=-=-=-
> -=-=-=-=-=-
>
> Thanks, will do more research on that part. By the way, Im trying to do
IVR
> where caller enter the pin the retrieve some information out of the MS
SQL.
>
Hi there,
I am wondering is there a preset command to saydecimal number? Currently if
you put comand in dialplan as SayNumber(1234) it will repeat to you. But how
about if the number is decimal like 12.34. Is there any command?
Thanks
--
Regards,
Sharon Lim
*Good memories are to be folded
Hi there, I am new to asterisk, I am trying to write a dialplan with 2 process running concurrently. Current dialplan only able to execute process with priority example exten =>100,1,Answer()exten =>100,2,Musiconhold()
exten =>100,3,Hangupis it possible to have process musiconhold/background and di
I am sorry cause i post this questions is not related to your problem, but i am having problem detecting my TDM400P which is a TDM400P problem. I manage to installed the card with compiling with zaptel and it got 2FXS and 2FXO.
I am having problem while reboot or restart the system. Kudzu seem to
Are you looking for an web interface that write to asterisk config files? if yes, you can look at freepbx.org . On 5/11/06,
Kerry Garrison <[EMAIL PROTECTED]> wrote:
You could install any number of interfaces but it does not
come with one.
Kerry
GarrisonDirector of Technical ServicesT
hi there, i am wondering can meetme.conf able to support diffferent context. Cause currently, it has [rooms] context. ]is it possible to have same conference number with different context?thanks
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Hmm...any idea where to define the context of a conference? Cause from my understanding, [rooms] context is a default. ThanksOn 5/18/06, Gavin Henry
<[EMAIL PROTECTED]> wrote:
> hi there,>> i am wondering can meetme.conf able to support diffferent context. Cause> currently, it has [rooms] context.
Hi there, Try use Web-MeetMe http://www.voip-info.org/wiki/view/MeetMe-Web-Control . I have tried to install but havent had time to configure. The latest version has a Conference schedulling..maybe this will helpwhen you get it working maybe can email me the configuration details...thanks..have
Hi there, SIP more commanly used and it is a openstandard. Meanwhile IAX2 is a protocol on asterisk. I dont think it will effect the cpu resources cause they are bid with the same codecs like G711 and etc..so if you used SIP or IAX2 it also refer to the same codecs...so dont think it will take a lo
Hi there,anyone in the community has manage to configure click to call features? Care to share.I have tried on this manual , seem got some software error like
http://www.voip-info.org/wiki/view/Asterisk+click+to+callSoftware error:
Unable to determine call statusMessage: Originate with 'Exten' requ
ne. I can post the .asp and .fla somewhere
if someone is interested in it.
-Original Message-----From: Sharon Lim
[mailto:[EMAIL PROTECTED]]Sent: Friday, June 09, 2006 6:37
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: [Asterisk-Users] click to call fe
fromthe web designer.http://www.snapanumber.com
On 6/9/06, Sharon Lim <[EMAIL PROTECTED]> wrote:> Hi colin,>> I am doing on php. But i would glad that you can share the codes as i will
> explore it.>> Thanks.>>> On 6/9/06, Colin Anderson <> [EMAIL PROTECTED]> wro
each ip phone need to register to be able to call the conference. Firstly, you need to create user with username and password under sip.conf, Then you need to create conference room which is meetme in meetme.conf
then you need to create extension to point to the conference room in extensions.confaf
yes. you can use xten http://www.xten.net/index.php?menu=download. free to download.On 6/15/06, Asterisk guy
<[EMAIL PROTECTED]> wrote:
are there any open source sip softphone (Window OS version )?___--Bandwidth and Colocation provided by Easynews.com --
Hi John, Your first question, I am not sure why but for this part i can explain abit
Also, on a side note, I have a context called [home] which each SIPPhone is associated with. Do I need to specify each extension inthere?SIP user can register as name as well . Doesnt means to have number. E
Did you install the sound packages such as mpg123-0.59r-1.i386.rpm ? Can download from http://rpm.pbone.net/index.php3/stat/4/idpl/516450/com/mpg123-0.59r-1.i386.rpm.html
good luck!On 6/16/06, kharris <[EMAIL PROTECTED]> wrote:
Can anyone point me in the direction for resources for troubleshooting
I had the same problem. I change some variable in zapata.conf such as in [defaults] context : 1. busydetect=yes2. busycount=43. hanguponpolarityswitch=yes ; some said need this variables4. rxgain=
1.0 5. txgain=1.0not sure which one effect it...but tried it...On 6/16/06, Steven Ringwald <[EMAIL PR
anyone have information on how the call back features work with asterisk? I means the dial plan or what so ever. thanks
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hi, Is it possible to have same sip context but refering to different username and context? Is it a must to have username(test) and the sip context [test] the same? what the different with username & sip context?
can username variable be alphanumeric, like using email address as username?default s
will ring which user phone. Cause if we have 2 jenny then if in extensions.conf exten=>200,1,Dial (SIP/jenny) which account will ring. That's my problem.
Thanks so much for the feedback. On 7/6/06, Olle E Johansson <[EMAIL PROTECTED]> wrote:
6 jul 2006 kl. 04.09 skrev Sharon Lim
Perhaps you can check on the dtmf code. On 7/14/06, Robert La Ferla <[EMAIL PROTECTED]> wrote:
When I dial out, I can't hear any ringing. I am using the latest SVNcode (SVN-branch-1.2-r37458M
). Is this a problem with Asterisk? Orwith my VOIP provider?_
Hi there, I would like to ask, is it possible to group sip user? Means group A with sip user 100,200 and group B with sip user 100,200? thanks in advance.
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Assuming, if 1 have 2 company and want to have same sip account context, how do i differentiate with it? Thanks in advance. On 7/17/06, El Flynn
<[EMAIL PROTECTED]> wrote:
Sharon Lim wrote:> Hi there,>> I would like to ask, is it possible to group sip user? Means group A with>
I have an old pbx and I want to pass callerid frm the old pbx to asterisk as a voicemail server. My old pbx have sent the callerid but i am not sure how to make it into dialpattern cause if I have 1000 callerid then i have to enter 1000 enter into
extensions.conf. I m using tdm400p where i pull an
ntrol it too be faster? Thanks very much!
On 8/1/06, Don Pobanz <[EMAIL PROTECTED]> wrote:
Sharon Liam wrote:> exten =>s, 1, Answer()> exten =>740,1,Voicemail(${EXTEN})>> After answer then it will get the callerid (I do a READ cmd then it read> as "s" exte
I am trying to do a simple agi connection to db with the guidance from http://www.voip-info.org/wiki/view/Asterisk+AGI+php Item 13 with ani.agi file, db and
extensions.conf13. another sample, ANI
Scenario - did callers call the Asterisks box and land on
the context did, Asterisks answers the cal
Did you leave any message in your voicemail? Cause by default if 10 seconds silence then it will end the recording. On 8/8/06, ismir saljic <
[EMAIL PROTECTED]> wrote:Hi , I have the problem with voicemail message
duration.Every message is only 10 seconds long.
just wanted to let you know yo
If i am not mistaken, you need to have another IAX user tables to store all the iax users. HEre is some example http://www.voip-info.org/wiki/view/Asterisk+RealTime+IAX
good luck!On 8/9/06, Shaun <[EMAIL PROTECTED]> wrote:
I'm attempting to setup asterisk running real-time with mysql. Right now Ic
wrote:
IAX is being read from the flat config like it
normally is. I can verify this because asterisk registers with my
provider.
-- ~Shaun
"Sharon Lim" <[EMAIL PROTECTED]> wrote
in message news:[EMAIL PROTECTED]
...If
i am not mistaken, you need to have another IA
http://www.itelbilling.com/ try this!
On 8/12/06, Wasif <[EMAIL PROTECTED]> wrote:
Hello,Does anyone know about open source wholesale billing for Asterisk?Thanks___--Bandwidth and Colocation provided by
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I am not sure whether username can be xyz.abc cause normally is single words. try to change it. On 8/14/06, Crazy Boy <
[EMAIL PROTECTED]> wrote:Hi,Thank you for your response. As you said, I executed the command "sip show registry". But, its not showing anything. Teliax people are also telling tha
is there something wrong with ur syntax at exten => _1XX,1,DIAL(SIP/teliax/${EXTEN},30,tr) exten => _1XX,1,DIAL(SIP/teliax,${EXTEN},30,tr)
On 8/14/06, Crazy Boy <
[EMAIL PROTECTED]> wrote:
Hi, My user name is : rudy.pandya Thank you.Sharon Lim <
[EMAIL PROTECTED]> wrote: I a
http://www.trixbox.org/modules/smartsection/item.php?itemid=4 On 8/28/06, Rizwan Hisham
<[EMAIL PROTECTED]> wrote:
hi guys,
i need to install the .tar.gz version of trixbox. i cant find any help files for installation in it and also there is no help for it on the website. can anybody please help?
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--asterisk-users mailing listTo UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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Hi all, I have tried to install freepbx and a2billing application. Now see both application is not integrated special on cdr part. Any idea how to integrated it?Confuse!-- Regards,
Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket
.. you may have better luck searching there.
bp
On 9/11/06, Steve Totaro <[EMAIL PROTECTED]
> wrote:
Sharon Lim wrote:> Hi all,>> I have tried to install freepbx and a2billing application. Now see
> both application is not integrated special on cdr part.>> Any ide
ns visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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asterisk-users mailing list
To UNSUB
Engineer
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the ba
my system. So plz help me uninstall trixbox.
And Sharon, thanx for the tip, but what about the rest of the scripts
trixbox has installed on my system. for example i dont want to start
asterisk on system startup, but trixbox does that. so anymore help will
be helpfull :)On 9/20/06, Tzafrir Cohen <
h Sunkara [EMAIL PROTECTED]
M:+91 9985077535O:+91 40 23114549F:+91 40 40208727
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focus?
Thanks-- Regards, Sharon Lim *Good memories are to be folded neatly and tucked away into the back pocket *
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wrote:
Sharon,
pbxware.bicomsystems.com
U: [EMAIL PROTECTED]
P: pbxware
All standard.
Steve
steve {at] bicomsystems [dot} com
- Original Message -
From:
Sharon Lim
To:
Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, September 28, 2006 9:52
terisk-users
-- Alex Robar[EMAIL PROTECTED]
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