[asterisk-users] meetme vs confbridge max user comparison wanted

2015-04-13 Thread Steve Edwards
I've got some Asterisk 11 (I could 'upgrade' if needed) hosts using meetme 
and I'd like to switch to confbridge to service more callers.


Can anyone reply with their experience along the lines of 'using meetme I 
was only getting x callers per server but with confbridge I now get y 
callers per server?'


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Re: [asterisk-users] Multicast to polycom from asterisk

2015-04-13 Thread Kevin Larsen
 I am using 11.17.0 -  and MulticastRTP. Doesnt seem to work with 
 polycom phones as other devices receive my multicast just fine.
 
 Is there something special to do to get multicast working with polycom 
phones?
 (other than enable multicast on the actual phone).

Didn't see if anyone had answered you or not on this, but Polycom uses 
their own form of MulticastRTP. It doesn't work with Asterisk's multicast 
setup. There is a company that makes a loud ringer/pager unit that can 
also be used to take in a sip call and multicast out to the Polycom 
phones. I haven't tried it myself as I just use the loud ringer 
capability, but it does appear that it would be a workable solution. I 
hesitate to promote the name here since this is non-commercial discussion, 
but let me know if you want to hear the actual product.-- 
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Re: [asterisk-users] Multicast to polycom from asterisk

2015-04-13 Thread Kevin Larsen
  I hesitate to promote the name here since this is non-commercial 
  discussion...
 
  but Polycom...
 
  Polycom phones...
 
 If mentioning Polycom is OK, I think mentioning a possible commercial 
 solution is OK.

In that case, the product in question is the Algo 8180 SIP Audio Alerter. 
I will state that I have not used this particular functionality, but it is 
mentioned in the users guide, so in theory you could use it as a bridge 
between Asterisk and mulitcast on the Polycom phones. YMMV.-- 
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Re: [asterisk-users] Multicast to polycom from asterisk

2015-04-13 Thread Steve Edwards

On Mon, 13 Apr 2015, Kevin Larsen wrote:

I hesitate to promote the name here since this is non-commercial 
discussion...



but Polycom...



Polycom phones...


If mentioning Polycom is OK, I think mentioning a possible commercial 
solution is OK.


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Re: [asterisk-users] I'm not able to install asterisk in AWS cloud

2015-04-13 Thread Mitul Limbani
Read the config.log file to know for which dependency it failed.
On 13-Apr-2015 2:08 PM, akhilesh chand omakhileshch...@gmail.com wrote:

 yes I called

 On Mon, Apr 13, 2015 at 1:27 PM, jg webaccounts...@jgoettgens.de wrote:



 I'm not able to install asterisk whenever I hit make command I get below
 error:

 make[1]: *** No rule to make target `../main/modules.link', needed by
 `asterisk'.  Stop.
 make: *** [main] Error 2


 Just guessing. Did you call ./configure?

 jg

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Re: [asterisk-users] I'm not able to install asterisk in AWS cloud

2015-04-13 Thread jg




I'm not able to install asterisk whenever I hit make command I get below error:

make[1]: *** No rule to make target `../main/modules.link', needed by 
`asterisk'.  Stop.
make: *** [main] Error 2



Just guessing. Did you call ./configure?

jg

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[asterisk-users] I'm not able to install asterisk in AWS cloud

2015-04-13 Thread akhilesh chand
Hi folks,


I'm not able to install asterisk whenever I hit make command I get below
error:

make[1]: *** No rule to make target `../main/modules.link', needed by
`asterisk'.  Stop.
make: *** [main] Error 2


Regards
Akhilesh
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Re: [asterisk-users] I'm not able to install asterisk in AWS cloud

2015-04-13 Thread akhilesh chand
yes I called

On Mon, Apr 13, 2015 at 1:27 PM, jg webaccounts...@jgoettgens.de wrote:



 I'm not able to install asterisk whenever I hit make command I get below
 error:

 make[1]: *** No rule to make target `../main/modules.link', needed by
 `asterisk'.  Stop.
 make: *** [main] Error 2


 Just guessing. Did you call ./configure?

 jg

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Re: [asterisk-users] I'm not able to install asterisk in AWS cloud

2015-04-13 Thread Guenther Boelter
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On 04/13/2015 03:40 PM, akhilesh chand wrote:
 Hi folks,
 
 
 I'm not able to install asterisk whenever I hit make command I get
 below error:
 
 make[1]: *** No rule to make target `../main/modules.link', needed
 by `asterisk'.  Stop. make: *** [main] Error 2
 
 
 Regards Akhilesh
 
 

Does `../main/modules.link' exist after running ./configure?

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ERPel, das deutsche Warenwirtschaftssystem fuer LINUX
http://www.davaosoft.com
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Re: [asterisk-users] I'm not able to install asterisk in AWS cloud

2015-04-13 Thread ajahar mohd
Hi Akhilesh,

Here is another fix,

getting the error, that: make[1]: *** No rule to make target
`../main/modules.link’, needed by `asterisk’. Stop. make: *** [main] Error
2 when compile asterisk

To get around this, just delete following line in file makeopts.embed_rules

EMBED_LDSCRIPTS+=../main/modules.link

Source: http://showmyroutes.com/wordpress/?p=500

Sincerely,

M azhar

http://www.nicacresults.com

On Mon, Apr 13, 2015 at 1:10 PM, akhilesh chand omakhileshch...@gmail.com
wrote:

 Hi folks,


 I'm not able to install asterisk whenever I hit make command I get below
 error:

 make[1]: *** No rule to make target `../main/modules.link', needed by
 `asterisk'.  Stop.
 make: *** [main] Error 2


 Regards
 Akhilesh

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Re: [asterisk-users] I'm not able to install asterisk in AWS cloud

2015-04-13 Thread akhilesh chand
yes modules.link is existing in pbx/modules.link.




On Mon, Apr 13, 2015 at 2:35 PM, Guenther Boelter gboel...@gmail.com
wrote:

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 On 04/13/2015 03:40 PM, akhilesh chand wrote:
  Hi folks,
 
 
  I'm not able to install asterisk whenever I hit make command I get
  below error:
 
  make[1]: *** No rule to make target `../main/modules.link', needed
  by `asterisk'.  Stop. make: *** [main] Error 2
 
 
  Regards Akhilesh
 
 

 Does `../main/modules.link' exist after running ./configure?

 - --
 DavaoSOFT, the home of ERPel
 ERPel, das deutsche Warenwirtschaftssystem fuer LINUX
 http://www.davaosoft.com
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Re: [asterisk-users] I'm not able to install asterisk in AWS cloud

2015-04-13 Thread ajahar mohd
The below error
make[1]: *** No rule to make target `../main/modules.link', needed by
`asterisk'.  Stop.
make: *** [main] Error 2
may be due problem in settings/configuration, double check these and try
again

On Mon, Apr 13, 2015 at 1:10 PM, akhilesh chand omakhileshch...@gmail.com
wrote:

 Hi folks,


 I'm not able to install asterisk whenever I hit make command I get below
 error:

 make[1]: *** No rule to make target `../main/modules.link', needed by
 `asterisk'.  Stop.
 make: *** [main] Error 2


 Regards
 Akhilesh

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[asterisk-users] Linking Asterisk 1.8 to late model Samsung PABX over PRI - transfer issues

2015-04-13 Thread Stefan Viljoen
Hi all

I've got a setup where I use a Sangoma PRI card driven via Sangoma WanPipe
to connect to a legacy Samsung PABX (I'm not sure which model) form Asterisk
1.8.11.0.

The reason is the customer has a large installed base of Samsung phones
physically connected to it and on each users desk. They did not want to
spring for a complete replace of all their Samsung phones with generic, and
dump their existing Samsung PABX (worth several hundred thousand in local
currency) for a 100% SIP phone + Asterisk setup.

They do want the calls recorded and accessed in a custom system, ergo
Asterisk - raison d'etre for the setup - it records and the 3rd party system
interfaces to it via the AMI to playback and record.

The Asterisk stands between the Samsung and the telco's SIP trunk, and
presents PRI NET to the Samsung, so it thinks it is directly connected to
the telco PRI trunk.

The Asterisk in turn connects to the telco --SIP-- trunk. (E. g. we protocol
translate on Asterisk level - it speaks PRI to the Samsung and SIP to the
telco)

Everything works fine, EXCEPT when users do transfers. If they Asterisk
transfer (#number) on the Samsung phones, it works fine, but ONLY if the
target Samsung phone is not busy with a conversation.

If the target Samsung phone IS busy with a conversation, that conversation
on the Samsung is cut off, and the person outside wanting to talk to the
consultant is also cut off - MOH ends and Asterisk emits standard CLI info
and runs the H extension on  the call, as it should for a normal hangup.

If they Samsung transfer (e. g. use the physical transfer button on the
Samsung handset) it works fine - if the user is busy, it rings back, if not,
the call goes through.

The reason they have to Asterisk transfer is that Asterisk can keep track
of which extension is actually linked to a call - this is vital for the 3rd
party system we have interrogating asterisk for extensions status for
business logic purposes over AMI.

If they Samsung transfer asterisk is oblivious of the new extension on
which the call is running, and therefore business logic breaks - it only has
that 1000 (reception) is busy, and has no inlking of the fact that, in the
Samsung, 1000 has transferred to 1023 (for example) and business logic
should run on 1023, NOT 1000...

Can anybody offer any comments?

Thanks

Stefan


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[asterisk-users] Multicast to polycom from asterisk

2015-04-13 Thread Jerry Geis
I am using 11.17.0 -  and MulticastRTP. Doesnt seem to work with
polycom phones as other devices receive my multicast just fine.

Is there something special to do to get multicast working with polycom
phones?
(other than enable multicast on the actual phone).

Thanks

Jerry
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Re: [asterisk-users] I'm not able to install asterisk in AWS cloud

2015-04-13 Thread Mitul Limbani
Did you do make menuselect and saved the options ?

Mitul
On 13-Apr-2015 5:43 PM, ajahar mohd azhar5...@gmail.com wrote:

 Hi Akhilesh,

 Here is another fix,

 getting the error, that: make[1]: *** No rule to make target
 `../main/modules.link’, needed by `asterisk’. Stop. make: *** [main] Error
 2 when compile asterisk

 To get around this, just delete following line in file makeopts.embed_rules

 EMBED_LDSCRIPTS+=../main/modules.link

 Source: http://showmyroutes.com/wordpress/?p=500

 Sincerely,

 M azhar

 http://www.nicacresults.com

 On Mon, Apr 13, 2015 at 1:10 PM, akhilesh chand omakhileshch...@gmail.com
  wrote:

 Hi folks,


 I'm not able to install asterisk whenever I hit make command I get below
 error:

 make[1]: *** No rule to make target `../main/modules.link', needed by
 `asterisk'.  Stop.
 make: *** [main] Error 2


 Regards
 Akhilesh

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[asterisk-users] error retrieving a video voicemail in asterisk 11

2015-04-13 Thread Steve Dolloff
Using asterisk 11.16.0 I am unable to retrieve any voicemail with a video 
attachment while using any video phone.  This does work in my 1.8.23.1 
installation.  The file is skipped with the ast_streamfile error (and moved to 
OLD), and the prompts following that message display the ast_best_codec error.

[Apr  7 16:05:50] WARNING[17497][C-6fdd]: file.c:1017 ast_streamfile: 
Unable to open /var/spool/asterisk/voicemail/default/2036/INBOX/msg (format 
(ulaw|h264)): No such file or directory [Apr  7 16:05:50] 
WARNING[17497][C-6fdd]: app_voicemail.c:8609 play_message: Playback of 
message /var/spool/asterisk/voicemail/default/2036/INBOX/msg failed
[Apr  7 16:05:50] -- SIP/2036-00ee Playing 'vm-advopts.gsm' (language 
'en')
[Apr  7 16:05:50] WARNING[17497][C-6fdd]: channel.c:940 ast_best_codec: 
Don't know any of (h264) formats

The file does exist in h264 format

-rw-r--r-- 1 root root 298102 Apr  7 16:05 msg.h264
-rw-r--r-- 1 root root301 Apr  7 16:05 msg.txt
-rw-r--r-- 1 root root 124524 Apr  7 16:05 msg.wav

Passthrough h264 video does work.  I do have h264 and ulaw codecs on the peer 
and videosupport=yes in sip.conf.  I also tried enabling h264 in the general 
section of sip.conf and gsm in voicemail.conf with the same results.

If I disable the h264 codec for the peer, I can listen to the audio portion of 
the message:

[Apr  7 16:41:05] -- SIP/2036-00f3 Playing 
'/var/spool/asterisk/voicemail/default/2036/Old/msg.slin' (language 'en')

Any guesses what I might be doing wrong?  Did something related change in 
asterisk 11?

-- Stephen


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Re: [asterisk-users] dial out with channel variable; sub-string usage

2015-04-13 Thread John Kiniston
BABY appears to be a global variable in your example.

In your CLI output testcarrier is a peer, It's not a variable at all.

The context field for your peer testcarrier is where incoming calls from
testcarrirer will be routed to.

Here is some example dialplan showing how you can use one context
'trunk1-out' to dial out through your trunk

[trunk1-out]
exten = _[0-9a-zA-Z].,1,Verbose(Dialing out TRUNK1)
same = n,Set(CHANNEL(amaflags)=BILLING)
same = n,Dial(PJSIP/${EXTEN}@trunk1,60,o)
same = n,Log(ERROR,Dial to ${EXTEN} ended - DIALSTATUS is ${DIALSTATUS} -
HANGUPCAUSE is ${HANGUPCAUSE})
same = n,Playtones(busy)
same = n,Busy(8)
same = n,Hangup()

[pstn-local]
exten = _976,1,Macro(fastbusy)
exten = _nxx,1,Goto(trunk1-out,${EXTEN},1)

[toll-free]
exten = _1855NXX,1,Goto(trunk1-out,${EXTEN},1)
exten = _1866NXX,1,Goto(trunk1-out,${EXTEN},1)
exten = _1877NXX,1,Goto(trunk1-out,${EXTEN},1)
exten = _1888NXX,1,Goto(trunk1-out,${EXTEN},1)
exten = _1800NXX,1,Goto(trunk1-out,${EXTEN},1)

[long-distance]
exten =
_1NXXNXX,1,Authenticate(/etc/asterisk//ldcodes.txt,a,KINISTON-001-)
same  =   n,Goto(trunk1-out,${EXTEN},1)

Then your internal phone context would include the above contexts:

[internal-phones]
include = pstn-local
include = toll-free
include = long-distance

Does that help make it clearer Thufir?


On Sun, Apr 12, 2015 at 5:31 PM, thufir hawat.thu...@gmail.com wrote:


 vici:~ # asterisk -rx sip show peers
 Name/username Host Dyn Forcerport ACL Port Status
 300/300   (Unspecified) D   N 0UNKNOWN
 301/301   192.168.0.24 D   N 5060 OK (29
 ms)
 302/302   (Unspecified) D   N 0UNKNOWN
 gs102/gs102   (Unspecified) D   N 0UNKNOWN
 testcarrier/19876543210 198.38.7.34  N
  5065 OK (82 ms)
 5 sip peers [Monitored: 2 online, 3 offline Unmonitored: 0 online, 0
 offline]
 vici:~ #
 vici:~ # asterisk -rx sip show peer testcarrier

   * Name   : testcarrier
   Secret   : Set
   MD5Secret: Not set
   Remote Secret: Not set
   Context  : default

 Which just brings me back to...the context for BABY...is...?  BABY is a
 channel variable.  In the above CLI output, the channel variable is
 testcarrier, which has a context of default.

 Each channel variable maps to at most one context?  Many channel variables
 can map to a single context?




A human being should be able to change a diaper, plan an invasion, butcher
a hog, conn a ship, design a building, write a sonnet, balance accounts,
build a wall, set a bone, comfort the dying, take orders, give orders,
cooperate, act alone, solve equations, analyze a new problem, pitch manure,
program a computer, cook a tasty meal, fight efficiently, die gallantly.
Specialization is for insects.
---Heinlein
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