[asterisk-users] Problem setting for incoming termination
Hi,
We have difficulty setting up the incoming termination for our
clients. Both the ends are using asterisk. The problem is unless we
use fromuser at client end, it does not work properly as expected.
Below is a configuration at our end. The problem is that whenever call
is received from the client, it goes to default context instead of
'dallas' context. Also, the ${CDR(accountcode)} variable remains
empty. Now, If we set fromuser field at the client end, then
everything starts working, however, in that case, it overrides the
callerid.
[dallas]
type=user
username=dallas
secret=somepassword
host=dynamic
nat=no
disallow=all
allow=g729
allow=ulaw
allow=alaw
accountcode=411
context=dallas
This is the configuration at client end.
[outgoing]
type=peer
username=dallas
secret=somepassword
host=
nat=no
disallow=all
allow=g729
allow=ulaw
allow=alaw
We do not require the client to register, neither we want them to use
fromuser field. I think we are doing some silly mistake since this
should be a simple configuration used by many. Please help
Thanks
Jim
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Re: [asterisk-users] BT killed Ribbit
we too had enough issues with ribbit support and moved to tringme for web based phone. On Wed, Aug 10, 2011 at 3:29 PM, Dean Collins wrote: > Any thoughts on why they did this? > > - > http://venturebeat.com/2011/08/09/bt-kills-ribbits-web-phone-platform-sends-customers-to-the-fast-growing-twilio/ > > > > what makes Twillio successful but another company willing to kill off a > $100m+ investment? > > > > > > Cheers, > > Dean > > > > > > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk+internal phones+recorded messages
You can have all this plus a lot more. What you need is configurations and dialplan code. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of neo haux Sent: Thursday, August 11, 2011 6:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk+internal phones+recorded messages Hi I want to change my old answering phone machine and two wireless phones with asterisk box + degium TDM400p (3 fxs+1FXO)+ one LAN phone(Aastra Nortel 9133i) + Wifi/SIP phone I am wondering if I´ll lost actual functionalities that are present in my old answering machine: 1) is it possible to show the caller number (coming from PSTN/FXO) in both SIP phones (wifi/SIP and LAN phone) ? Does SIP protocol take in charge this functionality 2) Most important question is : can I see on those internal phones (Wifi/SIP phone and LAN phone) that I´ve some recoded messages on asterisk. Indeed, I have this fucntionality with my old answering machine where I can see the number of new messages recorded in a big LCD screen. Thx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk reporting
On Thu, 2011-08-11 at 08:50 +0300, Richard Zulu wrote: > Hallo, > > > I have a production asterisk server running on Ubuntu however all my > configs where done using the CLI. > > > I would like to implement a reporting element into the server so I can > know the number of calls made, for what duration, on what dates. > > > What tool can I use that can fit within any already laid out dialplan? > > > Thanks > > > > Richard Zulu Hi All this information will be recorded in your CDR http://www.voip-info.org/wiki/view/Asterisk+cdr+csv You can also have this CDR stored into the database of your choice to make pulling reports from it easier. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trouble with *8 Pickup
We have a client that has sporadic problems with the *8 pickup facility. The server they are using is 1.8.5 and they are using Snom phones. Every now and then when they try to do a pickup from another phone they get a forbidden message on the phone and I can see the following in the logs. [Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL [Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL [Aug 8 11:51:53] WARNING[19314] chan_sip.c: No SIP tech_pvt! Fixup of SIP/-0404 failed. [Aug 8 11:51:53] WARNING[19314] channel.c: Fixup failed on channel SIP/-0404, strange things may happen. Does anyone know what this warning means? Thanks Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Background music during a call
Hi all,
Continuing a previous post regarding background music during a call,
there is a strange miserable problem which I am unable to understand.
It works fine on some systems as I have tested it with sip and with
dahdi + pri signalling with digium hardware on one of my production
server, but when I use it on my other production server running digium
hardware + ss7 signalling, it often looses music voice when muting
/unmuting or decreasing / increasing volume. and music never comes
back. some times music stops automatically after a couple of seconds
and does not resume by any mean.
below is the email chain for background music. any help?
--
[asterisk-users] Background music during a call
Rizwan Hisham rizwanhasham at gmail.com
Tue May 10 10:59:36 CDT 2011
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Ooops,
here is the correct version, Missed the capital X option in meetme before
which lets you control the volume etc.
[chat-room]
exten => love,1,Goto(love-a,1)
exten => love,2,Goto(love-b,1)
exten => love-a,1,Set(__MOH=love)
exten => love-a,n,Dial(Local/fake at chat-
room,,G(chat-room,chat,1))
exten => love-b,1,Goto(chat,100)
exten => curse,1,Goto(curse-a,1)
exten => curse,2,Goto(curse-b,1)
exten => curse-a,1,Set(__MOH=curse)
exten => curse-a,n,Dial(Local/fake at chat-room,,G(chat-room,chat,1))
exten => curse-b,1,Goto(chat,100)
exten => fake,1,Answer
exten => fake,2,MusicOnHold(${MOH})
exten => chat,1,Goto(100)
exten => chat,2,MeetMe(${MM},dx1qX)
exten => chat,100,MeetMe(${MM},daAx1qX)
exten => h,1,MeetMeAdmin(${MM},K)
exten => 4,1,MeetMeAdmin(${MM},t,2)
exten => 6,1,MeetMeAdmin(${MM},T,2)
exten => 2,1,MeetMeAdmin(${MM},M,2)
exten => 8,1,MeetMeAdmin(${MM},m,2)
exten=> _X,2,Goto(chat-room,chat,100)
On Tue, May 10, 2011 at 9:57 PM, Rizwan Hisham wrote:
> Very nice Loan. Here is the chat-room dialplan with a little tweek which
> lets you set the volume up/down or mute/unmute the song.
>
> Use 4/6 to increase/decrease the volume and 2/8 to mute/unmute the song
>
>
> [chat-room]
> exten => love,1,Goto(love-a,1)
> exten => love,2,Goto(love-b,1)
>
> exten => love-a,1,Set(__MOH=love)
> exten => love-a,n,Dial(Local/fake at chat-
> room,,G(chat-room,chat,1))
>
> exten => love-b,1,Goto(chat,100)
>
> exten => curse,1,Goto(curse-a,1)
> exten => curse,2,Goto(curse-b,1)
>
> exten => curse-a,1,Set(__MOH=curse)
> exten => curse-a,n,Dial(Local/fake at chat-room,,G(chat-room,chat,1))
>
> exten => curse-b,1,Goto(chat,100)
>
> exten => fake,1,Answer
> exten => fake,2,MusicOnHold(${MOH})
>
> exten => chat,1,Goto(100)
> exten => chat,2,MeetMe(${MM},dx1q)
>
> exten => chat,100,MeetMe(${MM},daAx1q)
>
> exten => h,1,MeetMeAdmin(${MM},K)
>
> exten => 4,1,MeetMeAdmin(${MM},t,2)
> exten => 6,1,MeetMeAdmin(${MM},T,2)
> exten => 2,1,MeetMeAdmin(${MM},M,2)
> exten => 8,1,MeetMeAdmin(${MM},m,2)
>
> exten=> _X,2,Goto(chat-room,chat,100)
>
> Here channel 2 always seem to be the one playing the MOH, thats why its
> hard coded into the MeetMeAdmin application.
>
> If there is a another way to know which channel is playing the song then
> please do let me know.
>
> Cheers
>
>
>
> On Tue, May 10, 2011 at 9:33 AM, Rizwan Hisham gmail.com>wrote:
>
>> Thanks a lot loan. Will try it today.
>>
>> Cheers
>>
>>
>> On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias wrote:
>>
>>> Updated dialplan: fix a typo when using MOH variable and now you have
>>> truly dynamic conference rooms.
>>>
>>> Have fun,
>>> Ioan.
>>>
>>> +
>>> exten => _[12]XXX,1,Set(__MM=${EPOCH})
>>> exten => _1XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,love,1))
>>> exten => _2XXX,n,Dial(SIP/${EXTEN:1},,G(chat-room,curse,1))
>>>
>>> [chat-room]
>>> exten => love,1,Goto(love-a,1)
>>> exten => love,2,Goto(love-b,1)
>>>
>>> exten => love-a,1,Set(__MOH=love)
>>> exten => love-a,n,Dial(Local/fake at chat-room,,G(chat-room,chat,1))
>>>
>>> exten => love-b,1,Goto(chat,100)
>>>
>>> exten => curse,1,Goto(curse-a,1)
>>> exten => curse,2,Goto(curse-b,1)
>>>
>>> exten => curse-a,1,Set(__MOH=curse)
>>> exten => curse-a,n,Dial(Local/fake at chat-room,,G(chat-room,chat,1))
>>>
>>> exten => curse-b,1,Goto(chat,100)
>>>
>>> exten => fake,1,Answer
>>> exten => fake,2,MusicOnHold(${MOH})
>>>
>>> exten => chat,1,Goto(100)
>>> exten => chat,2,MeetMe(${MM},dx1q)
>>>
>>> exten => chat,100,MeetMe(${MM},daAx1q)
>>>
>>> exten => h,1,MeetMeAdmin(${MM},K)
>>> +
>>>
>>> On Mon, May 9, 2011 at 4:02 PM, Ioan Indreias
>>> wrote:
>>> > I have tested the following dialplan and it could be used as a
>>> > starting point. What you have to resolve is how to generate different
>>> > MeetMe conference room - in the example we have only one room
Re: [asterisk-users] Problem setting for incoming termination
Anyone?
On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin wrote:
> Hi,
>
> We have difficulty setting up the incoming termination for our
> clients. Both the ends are using asterisk. The problem is unless we
> use fromuser at client end, it does not work properly as expected.
>
> Below is a configuration at our end. The problem is that whenever call
> is received from the client, it goes to default context instead of
> 'dallas' context. Also, the ${CDR(accountcode)} variable remains
> empty. Now, If we set fromuser field at the client end, then
> everything starts working, however, in that case, it overrides the
> callerid.
>
> [dallas]
> type=user
> username=dallas
> secret=somepassword
> host=dynamic
> nat=no
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> accountcode=411
> context=dallas
>
>
> This is the configuration at client end.
>
> [outgoing]
> type=peer
> username=dallas
> secret=somepassword
> host=
> nat=no
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
>
> We do not require the client to register, neither we want them to use
> fromuser field. I think we are doing some silly mistake since this
> should be a simple configuration used by many. Please help
>
> Thanks
> Jim
>
--
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Re: [asterisk-users] Problem setting for incoming termination
Best Regards,
Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
On Thu, Aug 11, 2011 at 3:09 PM, Jim Boykin wrote:
> Anyone?
>
> On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin wrote:
> > Hi,
> >
> > We have difficulty setting up the incoming termination for our
> > clients. Both the ends are using asterisk. The problem is unless we
> > use fromuser at client end, it does not work properly as expected.
> >
>
> Below is a configuration at our end. The problem is that whenever call
> > is received from the client, it goes to default context instead of
> > 'dallas' context. Also, the ${CDR(accountcode)} variable remains
> > empty. Now, If we set fromuser field at the client end, then
> > everything starts working, however, in that case, it overrides the
> > callerid.
> >
> > [dallas]
> > type=user
> > username=dallas
> > secret=somepassword
> > host=dynamic
> > nat=no
> > disallow=all
> > allow=g729
> > allow=ulaw
> > allow=alaw
> > accountcode=411
> > context=dallas
> >
> >
> > This is the configuration at client end.
> >
> > [outgoing]
> > type=peer
> > username=dallas
> > secret=somepassword
> > host=
> > nat=no
> > disallow=all
> > allow=g729
> > allow=ulaw
> > allow=alaw
> >
> > We do not require the client to register, neither we want them to use
> > fromuser field. I think we are doing some silly mistake since this
> > should be a simple configuration used by many. Please help
> >
> > Thanks
> > Jim
> >
>
how you connect this both server with over the internet or any ?
>
> --
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>
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Re: [asterisk-users] Problem setting for incoming termination
The problem seems like asterisk is not authenticating at all. It
accept the default invite and transfer it to default contact. ANy
help.
On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin wrote:
> Hi,
>
> We have difficulty setting up the incoming termination for our
> clients. Both the ends are using asterisk. The problem is unless we
> use fromuser at client end, it does not work properly as expected.
>
> Below is a configuration at our end. The problem is that whenever call
> is received from the client, it goes to default context instead of
> 'dallas' context. Also, the ${CDR(accountcode)} variable remains
> empty. Now, If we set fromuser field at the client end, then
> everything starts working, however, in that case, it overrides the
> callerid.
>
> [dallas]
> type=user
> username=dallas
> secret=somepassword
> host=dynamic
> nat=no
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
> accountcode=411
> context=dallas
>
>
> This is the configuration at client end.
>
> [outgoing]
> type=peer
> username=dallas
> secret=somepassword
> host=
> nat=no
> disallow=all
> allow=g729
> allow=ulaw
> allow=alaw
>
> We do not require the client to register, neither we want them to use
> fromuser field. I think we are doing some silly mistake since this
> should be a simple configuration used by many. Please help
>
> Thanks
> Jim
>
--
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Re: [asterisk-users] Problem setting for incoming termination
for start you could disable guest access in sip.conf, I guess you do not
need it
On 2011.08.11 14:29, Jim Boykin wrote:
> The problem seems like asterisk is not authenticating at all. It
> accept the default invite and transfer it to default contact. ANy
> help.
>
>
>
> On Thu, Aug 11, 2011 at 12:33 PM, Jim Boykin wrote:
>> Hi,
>>
>> We have difficulty setting up the incoming termination for our
>> clients. Both the ends are using asterisk. The problem is unless we
>> use fromuser at client end, it does not work properly as expected.
>>
>> Below is a configuration at our end. The problem is that whenever call
>> is received from the client, it goes to default context instead of
>> 'dallas' context. Also, the ${CDR(accountcode)} variable remains
>> empty. Now, If we set fromuser field at the client end, then
>> everything starts working, however, in that case, it overrides the
>> callerid.
>>
>> [dallas]
>> type=user
>> username=dallas
>> secret=somepassword
>> host=dynamic
>> nat=no
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>> accountcode=411
>> context=dallas
>>
>>
>> This is the configuration at client end.
>>
>> [outgoing]
>> type=peer
>> username=dallas
>> secret=somepassword
>> host=
>> nat=no
>> disallow=all
>> allow=g729
>> allow=ulaw
>> allow=alaw
>>
>> We do not require the client to register, neither we want them to use
>> fromuser field. I think we are doing some silly mistake since this
>> should be a simple configuration used by many. Please help
>>
>> Thanks
>> Jim
>>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://www.asterisk.org/hello
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>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Trouble with *8 Pickup
Ah, now this is interesting as one of our clients had the same problem the other day; in our case when they performed the *8 they got an extension unavailable from a completely different dialplan! This was on Asterisk 1.6 though with Snom phones. -- Thanks, Phil - Original Message - > We have a client that has sporadic problems with the *8 pickup > facility. > The server they are using is 1.8.5 and they are using Snom phones. > > Every now and then when they try to do a pickup from another phone > they > get a forbidden message on the phone and I can see the following in > the > logs. > > [Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL > [Aug 8 11:51:53] ERROR[19314] astobj2.c: user_data is NULL > [Aug 8 11:51:53] WARNING[19314] chan_sip.c: No SIP tech_pvt! Fixup > of SIP/-0404 failed. > [Aug 8 11:51:53] WARNING[19314] channel.c: Fixup failed on channel > SIP/-0404, strange things may happen. > > Does anyone know what this warning means? > > Thanks > > Ish > > -- > Ishfaq Malik > Software Developer > PackNet Ltd > > Office: 0161 660 3062 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with *8 Pickup
On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote: > Ah, now this is interesting as one of our clients had the same problem the > other day; in our case when they performed the *8 they got an extension > unavailable from a completely different dialplan! This was on Asterisk 1.6 > though with Snom phones. In the case of this server I was looking at, the only time this error occurred was when the pickup request happened in the same second as a dialplan step change so by the time the pick up of the channel was attempted, it no longer existed. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX Issues
Hi, > In my office I have 1000 ext, each users has it's own DID number. > What I would like is that each user can get a fax using his own number. I'm fighting with this since some time too. My experience is, that faxing over IAXModem is not reliable. So I'm going with T38Modem, my provider also supports T38 and finally I got Mail-2-Fax with T38Modem, HylaFax and Asterisk 1.4 runnig. I still have some issues with Fax-2-Mail, I had to patch T38Modem to get it working with Asterisk and it only works with Asterisk 1.8. But anyway I think it's better using T38Modem with T38 fax, than using IAXModem that uses G711 and which often makes problems with sending faxes. -- Chau y hasta luego, Thorolf -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FAX Issues
Thorolf Godawa wrote: My experience is, that faxing over IAXModem is not reliable. My experience is just the opposite, but I'm not faxing over IP. PRI, over slinear. Doug -- Ben Franklin quote: "Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety." -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with *8 Pickup
in my case i use snom 320 and 370 i flow this link and i can do the pichup with any issue http://asterisk.snom.com/index.php/Asterisk_1.4/Call_Pickup Regards. 2011/8/11 Ishfaq Malik > On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote: > > Ah, now this is interesting as one of our clients had the same problem > the other day; in our case when they performed the *8 they got an extension > unavailable from a completely different dialplan! This was on Asterisk 1.6 > though with Snom phones. > > In the case of this server I was looking at, the only time this error > occurred was when the pickup request happened in the same second as a > dialplan step change so by the time the pick up of the channel was > attempted, it no longer existed. > -- > Ishfaq Malik > Software Developer > PackNet Ltd > > Office: 0161 660 3062 > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with *8 Pickup
2011/8/11 Ishfaq Malik mailto:i...@pack-net.co.uk>> On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote: > Ah, now this is interesting as one of our clients had the same problem the other day; in our case when they performed the *8 they got an extension unavailable from a completely different dialplan! This was on Asterisk 1.6 though with Snom phones. In the case of this server I was looking at, the only time this error occurred was when the pickup request happened in the same second as a dialplan step change so by the time the pick up of the channel was attempted, it no longer existed. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 It's not just a snom/asterisk thing. I can replicate this with various phones and Asterisk 1.8.5. In fact with some phones the symptoms seemed worse where the phone *8 had been dialled on didn't hang up but thought it was on a call (while the caller had gone through to whatever the next dial plan priority was, a Queue in my test case). It makes perfect sense to me that a pickup should fail if your Dial has finished and * is stepping onto the next priority but a nicer Warning such as "Trying to pickup a non-existent channel" would be better. My test code was simply this: exten => 123321,1,Dial(SIP/5502,5) same => n,Answer same => n,Wait(1) same => n,Queue(booking,thHr) If you time the *8 just right so it is being handled during the end of the Dial then I got: [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL [Aug 11 16:26:18] WARNING[18458]: chan_sip.c:6429 sip_fixup: No SIP tech_pvt! Fixup of SIP/5501-01da failed. [Aug 11 16:26:18] WARNING[18458]: channel.c:6462 ast_do_masquerade: Fixup failed on channel SIP/5501-01da, strange things may happen. cheers, Paul. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any Method for capturing ISUP packets in DAHDI/ASTERISK
On Thu, Aug 11, 2011 at 11:57:46AM +0530, DHAVAL INDRODIYA wrote: > Hi All, > > I want packets [request/response] capture for ISUP packets , i have E1 line > terminated to my digium card > i just want a packets flow between my machine and teleco side, is any tool > or utility [command] availabele for > observation this packets and data. This issue and patch added pcap support for a guy who wanted to monitor ss7 traffic. You'll need to use dahdi 2.5. You'll also need to compile the "dahdi_pcap" program on your own, or write a script to exercise the DAHDI_RXMIRROR and DAHDI_TXMIRROR ioctl's. This is an unsupported interface. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any Method for capturing ISUP packets in DAHDI/ASTERISK
On Thu, Aug 11, 2011 at 12:43:43PM -0500, Russ Meyerriecks wrote: > On Thu, Aug 11, 2011 at 11:57:46AM +0530, DHAVAL INDRODIYA wrote: > > Hi All, > > > > I want packets [request/response] capture for ISUP packets , i have E1 line > > terminated to my digium card > > i just want a packets flow between my machine and teleco side, is any tool > > or utility [command] availabele for > > observation this packets and data. > > This issue and patch added pcap support for a guy who wanted to monitor > ss7 traffic. You'll need to use dahdi 2.5. You'll also need to compile > the "dahdi_pcap" program on your own, or write a script to exercise the > DAHDI_RXMIRROR and DAHDI_TXMIRROR ioctl's. This is an unsupported > interface. Forgot to link to the feature request: https://issues.asterisk.org/view.php?id=16831 -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Trouble with *8 Pickup
Paul Hayes writes: > If you time the *8 just right so it is being handled during the end of > the Dial then I got: > > [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data > is NULL > [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data > is NULL Does this happen when using the Pickup() application as well, or is it specific to *8? /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Where to proceed next
Hello list, I presently use the 1.4 releases because I enjoy sleeping at night. I understand that 1.4 reaches end-of-life in a little over 8 months (https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions). I also know (as best as I can) that no genie is going to make Asterisk 1.4 go "poof" on this date. My clients would probably sleep better thinking they were running a PBX that didn't have this "drop dead" date however. Since 1.6.X has the same time constraints as 1.4, it seems it would be a waste of time going that direction. Should I go down the 1.8 .X path to have 4 years of time, but the headaches that have been documented here, or pursue the 10.X which is presently considered Beta? (is it really beta, or just relabeled 1.8?). Thanks Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to proceed next
On Thu, Aug 11, 2011 at 3:04 PM, Danny Nicholas wrote: > Hello list, > > I presently use the 1.4 releases because I enjoy sleeping > at night. I understand that 1.4 reaches end-of-life in a little over 8 > months (https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions). I > also know (as best as I can) that no genie is going to make Asterisk 1.4 go > “poof” on this date. My clients would probably sleep better thinking they > were running a PBX that didn’t have this “drop dead” date however. Since > 1.6.X has the same time constraints as 1.4, it seems it would be a waste of > time going that direction. Should I go down the 1.8 .X path to have 4 years > of time, but the headaches that have been documented here, or pursue the > 10.X which is presently considered Beta? (is it really beta, or just > relabeled 1.8?). > > Thanks > > Danny Nicholas Regardless of what release you choose to use. The best thing you can do is to check that all the features and configurations you use are in the test-suite. Look at bamboo and see how the tests are going. If you have a feature in your dial-plan that concerns you, share it and think of a way to test this feature. -- ~ Andrew "lathama" Latham lath...@gmail.com http://lathama.net ~ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Where to proceed next
Does anybody ever update the software on the Panasonic phone system they had installed 30 years ago? Maybe if it ain't broke don't fix it. Hello list, I presently use the 1.4 releases because I enjoy sleeping at night. I understand that 1.4 reaches end-of-life in a little over 8 months (https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions). I also know (as best as I can) that no genie is going to make Asterisk 1.4 go "poof" on this date. My clients would probably sleep better thinking they were running a PBX that didn't have this "drop dead" date however. Since 1.6.X has the same time constraints as 1.4, it seems it would be a waste of time going that direction. Should I go down the 1.8 .X path to have 4 years of time, but the headaches that have been documented here, or pursue the 10.X which is presently considered Beta? (is it really beta, or just relabeled 1.8?). Thanks Danny Nicholas -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No more CDR record for simple Hangup?
On 11-08-08 09:50 AM, Miguel Molina wrote:
El 08/08/11 11:46, J Gao escribió:
On 11-08-06 10:06 AM, Miguel Molina wrote:
El 05/08/11 13:20, J Gao escribió:
I am using the new 1.8.5 and I just found out that Asterisk won't
record the call if the call just hangup. I did a test like this:
exten => 1009, 1, Hangup()
Then I called 1009:
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
-- Executing [1009@init-1005:1] Hangup("SIP/1007-003c", "")
in new stack
== Spawn extension (init-1005, 1009, 1) exited non-zero on
'SIP/1005-003c'
I am not sure why now Asterisk doesn't write this into CDR. In the
previous version Asterisk record the hangup call.
Is there anyway I can have the hangup write into the CDR?
What is your cdr.conf configuration?
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here is my cdr.conf:
==
[general]
endbeforehexten=yes
[csv]
usegmtime=yes; log date/time in GMT. Default is "no"
loguniqueid=yes ; log uniqueid. Default is "no"
loguserfield=yes ; log user field. Default is "no"
accountlogs=yes ; create separate log file for each account code.
Default is "yes"
Try this cdr.conf general setting: unanswered=yes
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Thank you for the help. It works after I enabled the unanswered=yes like
you suggested.
:)
Jian
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[asterisk-users] TLS Error on 1.6 and 1.8
Trying to setup UM with Office 365 which requires TLS. I've tried under 1.8.5.0 and under 1.6.2.16.1 and I get the same error: [Aug 11 06:50:20] VERBOSE[3023] tcptls.c: SSL certificate ok [Aug 11 06:50:20] VERBOSE[3023] tcptls.c: == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Aug 11 06:50:20] WARNING[3023] tcptls.c: FILE * open failed! Following the following two guides: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial http://www.msexchange.org/articles_tutorials/exchange-server-2010/mobility-client-access/how-configure-unified-messaging-asterisk-sip-gateway-part2.html Thanks-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] experiences sharing
Dear all may be it isn't related . but I want to shared my VOIP experiences in my new weblog. http://blog.lopl.net Help me to improve it by your comments and ideas. Best -- Pezhman Lali -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.5 - Ubuntu Pkg from diguim Repo - OPENH323 error
Hi Paul, Maybe you can give some help here: I'm trying to compile and build the debian source file of asterisk_1.8.5.0.orig.tar.gz and asterisk_1.8.5.0-1digium1~natty.debian.tar.gz. Howerver every time I'm trying to compile it, using ./configure of dpkg-buildpackage -rfakeroot -us -uc I get errors like this: checking for mandatory modules: CAP GSM OPENH323 IMAP_TK PWLIB... fail configure: *** configure: *** The OPENH323 installation appears to be missing or broken. configure: *** Either correct the installation, or run configure configure: *** including --without-h323. configure: *** configure: *** The PWLIB installation appears to be missing or broken. configure: *** Either correct the installation, or run configure configure: *** including --without-pwlib. make: *** [config.status] Error 1 dpkg-buildpackage: error: debian/rules build gave error exit status 2 root@FreePBX:/opt/asterisk/asterisk-1.8.5.0# I do have the dev files and all required pkg installed, but still no good. I manage u did compile it with these option on Can u maybe tell what u did? Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
