[asterisk-users] Anyone have a reliable T.38 Solution

[Matt Darnell]

Aloha,

We are looking to roll a solution that will have the following network layout:

ISDN-PRI -- Asterisk -- T.38 -- ATA -- Fax

Does version 1.8 with the Digium fax driver have this capability?  I
like 1.8 because it is a long term support version.

What ATA's are people using?

Any working solutions would be great!

Aloha,
Matt

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Lefteris Zafiris]

On 01/04/2012 07:51 AM, Bruce B wrote:
 And with recent version 14.3.2 I get:
 
 /usr/local/bin/sox FAIL formats: no handler for file extension `flac'
  -- speech-recog.agi: /usr/local/bin/sox failed: 512
 -- SIP/-002eAGI Script speech-recog.agi completed, returning 0
 
 Regards,
 
 
 On Wed, Jan 4, 2012 at 12:43 AM, Bruce B bruceb...@gmail.com wrote:
 
 Very interesting. I just tried to get it to work but it complains about
 sox. Probably you used a different version of sox?

 *PBX-*CLI /usr/bin/sox: invalid option -- -*
 */usr/bin/sox: invalid option -- n*
 */usr/bin/sox: invalid option -- o*
 */usr/bin/sox: -r must be given a positive integer*
 * -- speech-recog.agi: /usr/bin/sox failed: 512*

 I am using: *Package sox-12.18.1-1.el5_5.1.i386 *

 Thanks,



Note to self: Never release anything asterisk related without testing
on RHEL/Centos 5

Thank you for reporting this. I have replaced sox with flac and it seems
to work now on older platforms too (tested on Centos 5 with asterisk 1.4).
You can get the updated code here:
https://github.com/zaf/asterisk-speech-recog/tarball/master


Lefteris Zafiris

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[asterisk-users] Rami

[Arjan Kroon | Mobillion]

Hi,

Does anybody know if RAMI (Ruby Ami) is still functional?
And is this still compatible with asterisk 1.8

Best Regards,


Arjan Kroon
Mobillion BV

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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

[Jayesh Labade]

Please help me..

Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com



On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Hello Experts,

 I have pasted my issue in http://pastebin.com/zBGVmdcY

 I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08]
 NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to
 authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid
 ;tag=as57d3a806'
 i am unable to make outbound call from this trunk. while if i registered
 this trunk in softphone like Xlite, there is no problem with outbound
 calls. Help me.

 please find sip.conf file in http://pastebin.com/zBGVmdcY

 I have pasted sip debug with verbosity of failed call
 http://pastebin.com/jL2ki0s8


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com


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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

[virendra bhati]

Hi,

Give the complete details about the asterisk version, and SIP trunk conf
details


On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Please help me..

 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Hello Experts,

 I have pasted my issue in http://pastebin.com/zBGVmdcY

 I Cant able to Originate call from SIp trunk..I got this [Jan 3 11:52:08]
 NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to
 authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid
 ;tag=as57d3a806'
 i am unable to make outbound call from this trunk. while if i registered
 this trunk in softphone like Xlite, there is no problem with outbound
 calls. Help me.

 please find sip.conf file in http://pastebin.com/zBGVmdcY

 I have pasted sip debug with verbosity of failed call
 http://pastebin.com/jL2ki0s8


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



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Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

[Jayesh Labade]

Hi,

I am using asterisk ver 1.8.8.1.

My SIP trunk conf details are below..

[general]
context=default ; Default context for incoming calls
realm=192.168.1.55
allowguest=yes
realmauth=yes
send_rpid=pai

register = test02:test02@192.168.1.55


[test02]
type=peer
nat=no
canreinvite=no
host=192.168.1.55
;realm=test02@192.168.1.55
context=incoming
secret=test02
permit=192.168.1.0/255.255.255.0
username=test02
fromuser=test02
fromdomain=192.168.1.55
defaultuser=test02
insecure=invite,port
outboundproxy=192.168.1.55
promiscredir=yes
userphone=yes

For more details you can find my paste in pastebin.. Links given below.

While Dialing call fro Xlite send following Sip header F=
sip:test02@192.168.1.55. And if tried to register same account in asterisk
trunk i got F=sip:test02@anonymous.invalid in sip header. I dont know why
asterisk sends anonymous.invalid instead of domain name..Help me

Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com



On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.com wrote:

 Hi,

 Give the complete details about the asterisk version, and SIP trunk conf
 details


 On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Please help me..

 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade 
 jayesh.lab...@gmail.comwrote:

 Hello Experts,

 I have pasted my issue in http://pastebin.com/zBGVmdcY

 I Cant able to Originate call from SIp trunk..I got this [Jan 3
 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to
 authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid
 ;tag=as57d3a806'
 i am unable to make outbound call from this trunk. while if i registered
 this trunk in softphone like Xlite, there is no problem with outbound
 calls. Help me.

 please find sip.conf file in http://pastebin.com/zBGVmdcY

 I have pasted sip debug with verbosity of failed call
 http://pastebin.com/jL2ki0s8


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



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 --

 Thanks and regards

  Virendra Bhati
 +91-8885268942
 Software Engineer


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[asterisk-users] Asterisk won't start - trap invalid opcode

[Duncan Turnbull]

Hi there

Happy New Year

I have a new install of asterisk 1.8.8.1 on ubuntu server 
3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64 
x86_64 GNU/Linux

It has a Sangoma A200 card and I thought should be fairly standard but I have a 
new error when trying to start asterisk and I don't really know where to start

Initially asterisk was installed with dahdi from a package but sangoma didn't 
seem happy. Once I added dahdi from source sangoma wanpipe installed okay, but 
when I reloaded asterisk it stopped. So I removed all the packages (I believe I 
have but something could be hanging around) and rebuilt asterisk from source. 
Same errors. 

The only errors I can see are limited - I also stopped wan router and dahdi and 
I still get 
~# asterisk -cvv
Illegal instruction

Which isn't very informative. Kind of a fun challenge but not one I need right 
now

Google hasn't been able to find a similar issue 

My choices that I can see are:
- try another version of asterisk - delete everything and start again (which I 
thought I have tried but maybe not thorough enough)
- earlier version of ubuntu

But I would really like to understand what's clashing.

The sangoma card details are from lspci 

04:03.0 Network controller: Sangoma Technologies Corp. A200/Remora FXO/FXS 
Analog AFT card
Subsystem: NEC Corporation Device 0700
Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- 
Stepping- SERR- FastB2B- DisINTx-
Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- 
TAbort- MAbort- SERR- PERR- INTx-
Latency: 255 (1250ns min, 3750ns max)
Interrupt: pin A routed to IRQ 17
Region 0: Memory at dfdf (32-bit, non-prefetchable) [size=64K]

Output of dmesg looks fine until it gets to asterisk right at the bottom which 
gives this
[   13.419761] asterisk[1352] trap invalid opcode ip:5316f3 sp:7fff2db1a0f0 
error:0 in asterisk[40+1d6000]

dmesg long version

[5.051430] WANPIPE(tm) Hardware Support Module  3.5.24.0 (c) 1994-2010 
Sangoma Technologies Inc
[5.052051] usbcore: registered new interface driver sdlausb
[5.063329] dahdi: Telephony Interface Registered on major 196
[5.063334] dahdi: Version: 2.5.0.2
[5.071841] WANPIPE(tm) Interface Support Module 3.5.24.0 (c) 1994-2010 
Sangoma Technologies Inc
[5.098753] WANPIPE(tm) Multi-Protocol WAN Driver Module 3.5.24.0 (c) 
1994-2010 Sangoma Technologies Inc
[5.098759] wanpipe: Probing for WANPIPE hardware.
[5.098846] pci :04:03.0: PCI INT A - GSI 17 (level, low) - IRQ 17
[5.110126] wanpipe: AFT-A200-SH PCI FXO/FXS card found (HDLC rev.7), cpu(s) 
1, bus #4, slot #3, irq #5
[5.110137] wanpipe: Allocating maximum 1 devices: wanpipe1 - wanpipe1.
[5.110606] WANPIPE: TDM Codecs Initialized
[5.138361] WANPIPE(tm) Socket API Module 3.5.24.0 (c) 1994-2010 Sangoma 
Technologies Inc
[5.138368] NET: Registered protocol family 25
[5.171801] WANPIPE(tm) WANEC Layer 3.5.24.0 (c) 1995-2006 Sangoma 
Technologies Inc.
[5.171807] wanec_create_dev: Registering Wanpipe ECDEV Device!
[5.211707] wanpipe1: Starting WAN Setup
[5.211713] 
[5.211715] Processing WAN device wanpipe1...
[5.211719] wanpipe1: Locating: A200/A400/B600/B700/B800 card, CPU A, 
PciBus=4, PciSlot=3
[5.211728] wanpipe1: Found: A200/A400/B600/B700/B800 card, CPU A, PciBus=4, 
PciSlot=3, Port=0
[5.211785] wanpipe1: AFT PCI memory at 0xDFDF
[5.211788] wanpipe1: IRQ 17 allocated to the AFT PCI card
[5.211841] wanpipe1: Starting AFT Analog Hardware Init.
[5.211874] wanpipe1: Enabling front end link monitor
[5.211879] wanpipe1: Global Chip Configuration: used=1 used_type=1
[5.211906] wanpipe1: Global Front End Configuration!
[5.211909] wanpipe1: Configuring FXS/FXO Front End ...
[5.424637] wanpipe1: Module 1: Installed -- Auto FXO (AUSTRALIA mode)!
[5.624587] wanpipe1: Module 2: Installed -- Auto FXO (AUSTRALIA mode)!
[5.824637] wanpipe1: Module 3: Installed -- Auto FXO (AUSTRALIA mode)!
[6.025333] wanpipe1: Module 4: Installed -- Auto FXO (AUSTRALIA mode)!
[6.225169] wanpipe1: Module 5: Installed -- Auto FXO (AUSTRALIA mode)!
[6.425906] wanpipe1: Module 6: Installed -- Auto FXO (AUSTRALIA mode)!
[6.624559] wanpipe1: Module 7: Installed -- Auto FXO (AUSTRALIA mode)!
[6.824553] wanpipe1: Module 8: Installed -- Auto FXO (AUSTRALIA mode)!
[6.824560] wanpipe1: Running post initialization...
[6.824563] wanpipe1: Remora config done!
[6.824567] wanpipe1: AFT Data Mux Bit Map: 0x01234567
[6.824794] wanpipe1: Front End Interface Ready 0x
[6.824805] wanpipe1: Register EC interface wanec1 (usage 1, max ec chans 
32)!
[6.824811] wanpipe1: Configuring Device   :wanpipe1  FrmVr=07
[6.824814] wanpipe1:Global MTU = 1500
[6.824817] wanpipe1:Global MRU = 1500
[6.824819] wanpipe1:Data Mux Map   = 0x01234567
[6.824821] wanpipe1:

Re: [asterisk-users] Asterisk won't start - trap invalid opcode

[A J Stiles]

On Wednesday 04 January 2012, Duncan Turnbull wrote:
 Hi there
 
 Happy New Year
 
 I have a new install of asterisk 1.8.8.1 on ubuntu server
 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64
 x86_64 GNU/Linux
 
 It has a Sangoma A200 card and I thought should be fairly standard but I
 have a new error when trying to start asterisk and I don't really know
 where to start
 
 Initially asterisk was installed with dahdi from a package but sangoma
 didn't seem happy. Once I added dahdi from source sangoma wanpipe
 installed okay, but when I reloaded asterisk it stopped. So I removed all
 the packages (I believe I have but something could be hanging around) and
 rebuilt asterisk from source. Same errors.
 
 The only errors I can see are limited - I also stopped wan router and dahdi
 and I still get ~# asterisk -cvv
 Illegal instruction
 
 Which isn't very informative. Kind of a fun challenge but not one I need
 right now
 
 Google hasn't been able to find a similar issue

For what it's worth, I once tried installing Asterisk on an old VIA C7 box; 
and it turns out that this processor, while detecting as an i686, doesn't 
implement the full i686 instruction set -- and Asterisk is trying to use one 
of the non-implemented instructions.  Solution was to re-compile for i586.

It's just possible that something similar is going on here -- maybe your 
processor isn't implementing an instruction that Asterisk or Dahdi is relying 
on.  (It's my understanding that 64-bit processors don't fully implement the 
32-bit instructions when in 64-bit mode, but I wouldn't swear to that.)  Or 
maybe it's a library path problem -- something trying to use a 32-bit library 
instead of a 64-bit one, or vice versa.  Try ldd on the binaries.

What is your output from `cat /proc/cpuinfo` ?


If you have at least two SIP phones and/or an IAX route, try disabling Dahdi, 
and see if you can persuade Asterisk to run like that.  At least that should 
help track the problem down to one layer (Asterisk or Dahdi).

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Anyone have a reliable T.38 Solution

[David Klaverstyn]

I'm using  the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and 
rx_fax on multiple installations with no problems.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell
Sent: Wednesday, 4 January 2012 4:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Anyone have a reliable T.38 Solution

Aloha,

We are looking to roll a solution that will have the following network layout:

ISDN-PRI -- Asterisk -- T.38 -- ATA -- Fax

Does version 1.8 with the Digium fax driver have this capability?  I like 1.8 
because it is a long term support version.

What ATA's are people using?

Any working solutions would be great!

Aloha,
Matt

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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

[virendra bhati]

Hi checked your debug like.

Did you check that your SIP device ir registered with server ?
if yes then dial below command from CLI

*originate sip/test02 application dial*



On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Hi,

 I am using asterisk ver 1.8.8.1.

 My SIP trunk conf details are below..

 [general]
 context=default ; Default context for incoming calls
 realm=192.168.1.55
 allowguest=yes
 realmauth=yes
 send_rpid=pai

 register = test02:test02@192.168.1.55


 [test02]
 type=peer
 nat=no
 canreinvite=no
 host=192.168.1.55
 ;realm=test02@192.168.1.55
 context=incoming
 secret=test02
 permit=192.168.1.0/255.255.255.0
 username=test02
 fromuser=test02
 fromdomain=192.168.1.55
 defaultuser=test02
 insecure=invite,port
 outboundproxy=192.168.1.55
 promiscredir=yes
 userphone=yes

 For more details you can find my paste in pastebin.. Links given below.

 While Dialing call fro Xlite send following Sip header F=
 sip:test02@192.168.1.55. And if tried to register same account in
 asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I dont
 know why asterisk sends anonymous.invalid instead of domain name..Help me


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.com wrote:

 Hi,

 Give the complete details about the asterisk version, and SIP trunk conf
 details


 On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Please help me..

 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade 
 jayesh.lab...@gmail.comwrote:

 Hello Experts,

 I have pasted my issue in http://pastebin.com/zBGVmdcY

 I Cant able to Originate call from SIp trunk..I got this [Jan 3
 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed to
 authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid
 ;tag=as57d3a806'
 i am unable to make outbound call from this trunk. while if i
 registered this trunk in softphone like Xlite, there is no problem with
 outbound calls. Help me.

 please find sip.conf file in http://pastebin.com/zBGVmdcY

 I have pasted sip debug with verbosity of failed call
 http://pastebin.com/jL2ki0s8


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 --
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 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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   http://lists.digium.com/mailman/listinfo/asterisk-users




 --

 Thanks and regards

  Virendra Bhati
 +91-8885268942
 Software Engineer


 --
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 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users



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-- 

Thanks and regards

 Virendra Bhati
+91-8885268942
Software Engineer
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Re: [asterisk-users] Asterisk won't start - trap invalid opcode

[Duncan Turnbull]

On 4/01/2012, at 11:47 PM, A J Stiles wrote:


 
 For what it's worth, I once tried installing Asterisk on an old VIA C7 box; 
 and it turns out that this processor, while detecting as an i686, doesn't 
 implement the full i686 instruction set -- and Asterisk is trying to use one 
 of the non-implemented instructions.  Solution was to re-compile for i586.
 

Thanks very much AJ

That did appear as one of the few google comments I found but I couldn't figure 
out whether it applies 

The outputs are below if you can interpret them, I can see lm in the cpu proc 
info but don't know how to check for better compatibility

 It's just possible that something similar is going on here -- maybe your 
 processor isn't implementing an instruction that Asterisk or Dahdi is relying 
 on.  (It's my understanding that 64-bit processors don't fully implement the 
 32-bit instructions when in 64-bit mode, but I wouldn't swear to that.)  Or 
 maybe it's a library path problem -- something trying to use a 32-bit library 
 instead of a 64-bit one, or vice versa.  Try ldd on the binaries.
 
ldd -v /usr/sbin/asterisk 
linux-vdso.so.1 =  (0x7fff407ff000)
libssl.so.1.0.0 = /lib/x86_64-linux-gnu/libssl.so.1.0.0 
(0x7f72accc)
libcrypto.so.1.0.0 = /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 
(0x7f72ac911000)
libc.so.6 = /lib/x86_64-linux-gnu/libc.so.6 (0x7f72ac571000)
libxml2.so.2 = /usr/lib/libxml2.so.2 (0x7f72ac216000)
libdl.so.2 = /lib/x86_64-linux-gnu/libdl.so.2 (0x7f72ac012000)
libpthread.so.0 = /lib/x86_64-linux-gnu/libpthread.so.0 
(0x7f72abdf4000)
libtinfo.so.5 = /lib/libtinfo.so.5 (0x7f72abbcd000)
libm.so.6 = /lib/x86_64-linux-gnu/libm.so.6 (0x7f72ab949000)
libresolv.so.2 = /lib/x86_64-linux-gnu/libresolv.so.2 
(0x7f72ab72d000)
libz.so.1 = /lib/x86_64-linux-gnu/libz.so.1 (0x7f72ab515000)
/lib64/ld-linux-x86-64.so.2 (0x7f72acf1b000)

Version information:
/usr/sbin/asterisk:
libdl.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libdl.so.2
libresolv.so.2 (GLIBC_2.2.5) = 
/lib/x86_64-linux-gnu/libresolv.so.2
libxml2.so.2 (LIBXML2_2.6.0) = /usr/lib/libxml2.so.2
libxml2.so.2 (LIBXML2_2.4.30) = /usr/lib/libxml2.so.2
libcrypto.so.1.0.0 (OPENSSL_1.0.0) = 
/lib/x86_64-linux-gnu/libcrypto.so.1.0.0
libm.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libm.so.6
libpthread.so.0 (GLIBC_2.3.3) = 
/lib/x86_64-linux-gnu/libpthread.so.0
libpthread.so.0 (GLIBC_2.2.5) = 
/lib/x86_64-linux-gnu/libpthread.so.0
libc.so.6 (GLIBC_2.8) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.3) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.3.2) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6
libssl.so.1.0.0 (OPENSSL_1.0.0) = 
/lib/x86_64-linux-gnu/libssl.so.1.0.0
/lib/x86_64-linux-gnu/libssl.so.1.0.0:
libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6
libcrypto.so.1.0.0 (OPENSSL_1.0.0) = 
/lib/x86_64-linux-gnu/libcrypto.so.1.0.0
/lib/x86_64-linux-gnu/libcrypto.so.1.0.0:
libdl.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libdl.so.2
libc.so.6 (GLIBC_2.3) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.7) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6
/lib/x86_64-linux-gnu/libc.so.6:
ld-linux-x86-64.so.2 (GLIBC_2.3) = /lib64/ld-linux-x86-64.so.2
ld-linux-x86-64.so.2 (GLIBC_PRIVATE) = 
/lib64/ld-linux-x86-64.so.2
/usr/lib/libxml2.so.2:
libz.so.1 (ZLIB_1.2.2.3) = /lib/x86_64-linux-gnu/libz.so.1
libdl.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libdl.so.2
libm.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libm.so.6
libc.so.6 (GLIBC_2.7) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.3.2) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6
libc.so.6 (GLIBC_2.3) = /lib/x86_64-linux-gnu/libc.so.6

Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

[Jayesh Labade]

Hi virendra,

Dialed same command.. I got below output

ast18*CLI originate sip/test02 application dial
  == Using SIP RTP CoS mark 5
[Jan  4 14:13:07] NOTICE[29823]: chan_sip.c:19718 handle_response_invite:
Failed to authenticate on INVITE to 'Anonymous
sip:test02@anonymous.invalid:192;tag=as417a5527'


Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com



On Wed, Jan 4, 2012 at 4:35 PM, virendra bhati virbh...@gmail.com wrote:

 Hi checked your debug like.

 Did you check that your SIP device ir registered with server ?
 if yes then dial below command from CLI

 *originate sip/test02 application dial*




 On Wed, Jan 4, 2012 at 3:33 PM, Jayesh Labade jayesh.lab...@gmail.comwrote:

 Hi,

 I am using asterisk ver 1.8.8.1.

 My SIP trunk conf details are below..

 [general]
 context=default ; Default context for incoming calls
 realm=192.168.1.55
 allowguest=yes
 realmauth=yes
 send_rpid=pai

 register = test02:test02@192.168.1.55


 [test02]
 type=peer
 nat=no
 canreinvite=no
 host=192.168.1.55
 ;realm=test02@192.168.1.55
 context=incoming
 secret=test02
 permit=192.168.1.0/255.255.255.0
 username=test02
 fromuser=test02
 fromdomain=192.168.1.55
 defaultuser=test02
 insecure=invite,port
 outboundproxy=192.168.1.55
 promiscredir=yes
 userphone=yes

 For more details you can find my paste in pastebin.. Links given below.

 While Dialing call fro Xlite send following Sip header F=
 sip:test02@192.168.1.55. And if tried to register same account in
 asterisk trunk i got F=sip:test02@anonymous.invalid in sip header. I
 dont know why asterisk sends anonymous.invalid instead of domain name..Help
 me


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 3:16 PM, virendra bhati virbh...@gmail.comwrote:

 Hi,

 Give the complete details about the asterisk version, and SIP trunk conf
 details


 On Wed, Jan 4, 2012 at 3:07 PM, Jayesh Labade 
 jayesh.lab...@gmail.comwrote:

 Please help me..

 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



 On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com
  wrote:

 Hello Experts,

 I have pasted my issue in http://pastebin.com/zBGVmdcY

 I Cant able to Originate call from SIp trunk..I got this [Jan 3
 11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite: Failed 
 to
 authenticate on INVITE to 'Anonymous sip:test02@anonymous.invalid
 ;tag=as57d3a806'
 i am unable to make outbound call from this trunk. while if i
 registered this trunk in softphone like Xlite, there is no problem with
 outbound calls. Help me.

 please find sip.conf file in http://pastebin.com/zBGVmdcY

 I have pasted sip debug with verbosity of failed call
 http://pastebin.com/jL2ki0s8


 Best Regards,
 *Jayesh Labade*
 e-mail: jayesh.lab...@gmail.com



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 Thanks and regards

  Virendra Bhati
 +91-8885268942
 Software Engineer


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 Thanks and regards

  Virendra Bhati
 +91-8885268942
 Software Engineer


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[asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like 
my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
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Re: [asterisk-users] Asterisk won't start - trap invalid opcode

[Duncan Turnbull]

I loaded the latest 1.6 which gets slightly further and a core dump shows this, 
but its past my ability to interpret

# gdb -se asterisk -c core | tee /tmp/backtrace.txt
GNU gdb (Ubuntu/Linaro 7.3-0ubuntu2) 7.3-2011.08
Copyright (C) 2011 Free Software Foundation, Inc.
License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html
This is free software: you are free to change and redistribute it.
There is NO WARRANTY, to the extent permitted by law.  Type show copying
and show warranty for details.
This GDB was configured as x86_64-linux-gnu.
For bug reporting instructions, please see:
http://bugs.launchpad.net/gdb-linaro/...
Reading symbols from /usr/sbin/asterisk...done.
[New LWP 19322]
[New LWP 19323]
[New LWP 19324]
[New LWP 19325]
[New LWP 19326]

warning: Can't read pathname for load map: Input/output error.
[Thread debugging using libthread_db enabled]
Core was generated by `asterisk -d -g -cvvv'.
Program terminated with signal 4, Illegal instruction.
#0  0x00500eab in tzload (name=optimized out, sp=0x1fc7950, 
doextend=1) at stdtime/localtime.c:424
424 static int tzload(const char *name, struct state * const sp, const int 
doextend)



On 5/01/2012, at 12:13 AM, Duncan Turnbull wrote:

 On 4/01/2012, at 11:47 PM, A J Stiles wrote:
 
 
 
 For what it's worth, I once tried installing Asterisk on an old VIA C7 box; 
 and it turns out that this processor, while detecting as an i686, doesn't 
 implement the full i686 instruction set -- and Asterisk is trying to use one 
 of the non-implemented instructions.  Solution was to re-compile for i586.
 
 
 Thanks very much AJ
 
 That did appear as one of the few google comments I found but I couldn't 
 figure out whether it applies 
 
 The outputs are below if you can interpret them, I can see lm in the cpu proc 
 info but don't know how to check for better compatibility
 
 It's just possible that something similar is going on here -- maybe your 
 processor isn't implementing an instruction that Asterisk or Dahdi is 
 relying 
 on.  (It's my understanding that 64-bit processors don't fully implement the 
 32-bit instructions when in 64-bit mode, but I wouldn't swear to that.)  Or 
 maybe it's a library path problem -- something trying to use a 32-bit 
 library 
 instead of a 64-bit one, or vice versa.  Try ldd on the binaries.
 
 ldd -v /usr/sbin/asterisk 
   linux-vdso.so.1 =  (0x7fff407ff000)
   libssl.so.1.0.0 = /lib/x86_64-linux-gnu/libssl.so.1.0.0 
 (0x7f72accc)
   libcrypto.so.1.0.0 = /lib/x86_64-linux-gnu/libcrypto.so.1.0.0 
 (0x7f72ac911000)
   libc.so.6 = /lib/x86_64-linux-gnu/libc.so.6 (0x7f72ac571000)
   libxml2.so.2 = /usr/lib/libxml2.so.2 (0x7f72ac216000)
   libdl.so.2 = /lib/x86_64-linux-gnu/libdl.so.2 (0x7f72ac012000)
   libpthread.so.0 = /lib/x86_64-linux-gnu/libpthread.so.0 
 (0x7f72abdf4000)
   libtinfo.so.5 = /lib/libtinfo.so.5 (0x7f72abbcd000)
   libm.so.6 = /lib/x86_64-linux-gnu/libm.so.6 (0x7f72ab949000)
   libresolv.so.2 = /lib/x86_64-linux-gnu/libresolv.so.2 
 (0x7f72ab72d000)
   libz.so.1 = /lib/x86_64-linux-gnu/libz.so.1 (0x7f72ab515000)
   /lib64/ld-linux-x86-64.so.2 (0x7f72acf1b000)
 
   Version information:
   /usr/sbin/asterisk:
   libdl.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libdl.so.2
   libresolv.so.2 (GLIBC_2.2.5) = 
 /lib/x86_64-linux-gnu/libresolv.so.2
   libxml2.so.2 (LIBXML2_2.6.0) = /usr/lib/libxml2.so.2
   libxml2.so.2 (LIBXML2_2.4.30) = /usr/lib/libxml2.so.2
   libcrypto.so.1.0.0 (OPENSSL_1.0.0) = 
 /lib/x86_64-linux-gnu/libcrypto.so.1.0.0
   libm.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libm.so.6
   libpthread.so.0 (GLIBC_2.3.3) = 
 /lib/x86_64-linux-gnu/libpthread.so.0
   libpthread.so.0 (GLIBC_2.2.5) = 
 /lib/x86_64-linux-gnu/libpthread.so.0
   libc.so.6 (GLIBC_2.8) = /lib/x86_64-linux-gnu/libc.so.6
   libc.so.6 (GLIBC_2.3) = /lib/x86_64-linux-gnu/libc.so.6
   libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6
   libc.so.6 (GLIBC_2.3.2) = /lib/x86_64-linux-gnu/libc.so.6
   libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6
   libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6
   libssl.so.1.0.0 (OPENSSL_1.0.0) = 
 /lib/x86_64-linux-gnu/libssl.so.1.0.0
   /lib/x86_64-linux-gnu/libssl.so.1.0.0:
   libc.so.6 (GLIBC_2.4) = /lib/x86_64-linux-gnu/libc.so.6
   libc.so.6 (GLIBC_2.3.4) = /lib/x86_64-linux-gnu/libc.so.6
   libc.so.6 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libc.so.6
   libcrypto.so.1.0.0 (OPENSSL_1.0.0) = 
 /lib/x86_64-linux-gnu/libcrypto.so.1.0.0
   /lib/x86_64-linux-gnu/libcrypto.so.1.0.0:
   libdl.so.2 (GLIBC_2.2.5) = /lib/x86_64-linux-gnu/libdl.so.2
   libc.so.6 (GLIBC_2.3) = 

Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Julian Lyndon-Smith]

this looks great - is there any chance of coverting the googletts.agi
to use flac as well ?

Julian

On 4 January 2012 09:06, Lefteris Zafiris zaf@gmail.com wrote:
 On 01/04/2012 07:51 AM, Bruce B wrote:
 And with recent version 14.3.2 I get:

 /usr/local/bin/sox FAIL formats: no handler for file extension `flac'
  -- speech-recog.agi: /usr/local/bin/sox failed: 512
     -- SIP/-002eAGI Script speech-recog.agi completed, returning 0

 Regards,


 On Wed, Jan 4, 2012 at 12:43 AM, Bruce B bruceb...@gmail.com wrote:

 Very interesting. I just tried to get it to work but it complains about
 sox. Probably you used a different version of sox?

 *PBX-*CLI /usr/bin/sox: invalid option -- -*
 */usr/bin/sox: invalid option -- n*
 */usr/bin/sox: invalid option -- o*
 */usr/bin/sox: -r must be given a positive integer*
 * -- speech-recog.agi: /usr/bin/sox failed: 512*

 I am using: *Package sox-12.18.1-1.el5_5.1.i386 *

 Thanks,



 Note to self: Never release anything asterisk related without testing
 on RHEL/Centos 5

 Thank you for reporting this. I have replaced sox with flac and it seems
 to work now on older platforms too (tested on Centos 5 with asterisk 1.4).
 You can get the updated code here:
 https://github.com/zaf/asterisk-speech-recog/tarball/master

 
 Lefteris Zafiris

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-- 
Julian Lyndon-Smith
IT Director, Dot R Limited

I don’t care if it works on your machine!  We are not shipping your machine!”

The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Lefteris Zafiris]

On 01/04/2012 04:07 PM, Julian Lyndon-Smith wrote:
 this looks great - is there any chance of coverting the googletts.agi
 to use flac as well ?
 
 Julian
 

In googletts.agi we get the voice data from google in mp3 and we convert
it in a format that asterisk can read and playback (slin). If we store it
in flac asterisk wont be able to read it natively and we would have to
convert it each time we want to play it back to the user.

In the speech recognition script we have to convert the voice data in
flac before sending it to google because that's the accepted format.

Is there some particular reason you want the googletts.agi data in flac?


Lefteris Zafiris



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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Julian Lyndon-Smith]

the only reason is that I didn't want to have to install sox. Lazy.
that's all ;) Just another piece of software to find and install

running on amazon ec2, is the best thing to download the source and
compile sox ?

Thanks

Julian



On 4 January 2012 14:18, Lefteris Zafiris zaf@gmail.com wrote:
 On 01/04/2012 04:07 PM, Julian Lyndon-Smith wrote:
 this looks great - is there any chance of coverting the googletts.agi
 to use flac as well ?

 Julian


 In googletts.agi we get the voice data from google in mp3 and we convert
 it in a format that asterisk can read and playback (slin). If we store it
 in flac asterisk wont be able to read it natively and we would have to
 convert it each time we want to play it back to the user.

 In the speech recognition script we have to convert the voice data in
 flac before sending it to google because that's the accepted format.

 Is there some particular reason you want the googletts.agi data in flac?

 
 Lefteris Zafiris



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-- 
Julian Lyndon-Smith
IT Director, Dot R Limited

I don’t care if it works on your machine!  We are not shipping your machine!”

The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Lefteris Zafiris]

On 01/04/2012 04:24 PM, Julian Lyndon-Smith wrote:
 the only reason is that I didn't want to have to install sox. Lazy.
 that's all ;) Just another piece of software to find and install
 
 running on amazon ec2, is the best thing to download the source and
 compile sox ?
 
 Thanks
 

It should be on your distro repos already.


Lefteris Zafiris

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Julian Lyndon-Smith]

nope :(

On 4 January 2012 14:29, Lefteris Zafiris zaf@gmail.com wrote:
 On 01/04/2012 04:24 PM, Julian Lyndon-Smith wrote:
 the only reason is that I didn't want to have to install sox. Lazy.
 that's all ;) Just another piece of software to find and install

 running on amazon ec2, is the best thing to download the source and
 compile sox ?

 Thanks


 It should be on your distro repos already.

 
 Lefteris Zafiris

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IT Director, Dot R Limited

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The kangaroo dances: http://www.youtube.com/watch?v=MAWl5iYOaUg

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[asterisk-users] Which QSIG variant and profiles does asterisk support ?

[Olivier]

Hello,

Which QSIG (ECMA or ISO) variant and profiles does asterisk support ?
(I could not find this info inside chan_dahdi.conf)

Regards

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Re: [asterisk-users] Rami

[gokulnath]

Hey,
There is a new kid in town if you want to code in ruby. Use
adhearsionhttps://github.com/adhearsion/adhearsion/wiki,
it's a framework to make voice apps.

On Wed, Jan 4, 2012 at 2:49 PM, Arjan Kroon | Mobillion 
arjan.kr...@mobillion.nl wrote:

 Hi,

 Does anybody know if RAMI (Ruby Ami) is still functional?
 And is this still compatible with asterisk 1.8

 Best Regards,


 Arjan Kroon
 Mobillion BV

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Thanks  Regards
Gokulnath
@8129845320
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[asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)

[Ahmed Munir]

Hi all,

I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
Currently my AGI is working fine in my two servers but not in my other four
servers. When  I tried execute an AGI (as a user asterisk) in command line
it works fine (even I also declare environmental variables in user profile
and in my AGI), but when I tried to call my AGI (perl) in dial plan, it
don't get executed.

Please advise me to resolve this issue.

-- 
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Problem w/ PC port on Polycom 335

[Mike Diehl]

We did get this fixed.  Turns out that my tech didn't reboot the phone after 
disabling the vlan configuration.  He's new and still learning. 

Thank you for your time and suggestions.

On Monday 02 January 2012 6:04:49 pm Jim DeVito wrote:
 Agreed. Check the switch for some kind of port security. Most of the time
 this would disable the interface if more than one MAC is present but you
 never know. Are there blinky lights on the pc?
 
 Also if provisioning via some sort of server check the MAC-boot log that
 the pgone uploads.
 
 Good Luck!!
 
 Thanks!!
 
 Jim.
 
 - Original message -
 
  Mike Diehl wrote:
   Usually, it just works...
   
   Any ideas?
  
  I've seen this before.
  
  One of our facilities have 'smart or managed' switches that have caused
  no ends of problems, including preventing computers plugged into the
  phones not having network access.
  
  You may want to review your switches.
  
  Doug
 
http://www.asterisk.org/hello
 
  asterisk-users mailing list
 
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Take care and have fun,
Mike Diehl.

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Re: [asterisk-users] Set Call Codec in extension.conf

[Eric Wieling]

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call only 
cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like 
my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
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Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call only 
cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like 
my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
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Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video its 
sent to the other client as video call .I tried settings the 
SIP_CODED_INBOUND and outbound also ... but no luck

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling 
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call only 
cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like 
my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
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Re: [asterisk-users] Set Call Codec in extension.conf

[Eric Wieling]

1.6 does not support setting the outbound codec.1.8 uses different 
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source for 
the 1.8 information,.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video its 
sent to the other client as video call .I tried settings the 
SIP_CODED_INBOUND and outbound also ... but no luck 

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling 
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call only 
cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like 
my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
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Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

I tried also in asterisk 1.8 setting outbound variable  but didnt work also 

https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried  but still I get a video call

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling 
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 does not support setting the outbound codec.1.8 uses different 
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source for 
the 1.8 information,.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video its 
sent to the other client as video call .I tried settings the 
SIP_CODED_INBOUND and outbound also ... but no luck 

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling 
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call only 
cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its like 
my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
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Re: [asterisk-users] Set Call Codec in extension.conf

[Danny Nicholas]

My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I tried also in asterisk 1.8 setting outbound variable  but didnt work
also 
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried  but still I get a video call

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 does not support setting the outbound codec.1.8 uses different
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source
for the 1.8 information,.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video
its sent to the other client as video call .I tried settings the
SIP_CODED_INBOUND and outbound also ... but no luck

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call
only cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its
like my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
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Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)

[Danny Nicholas]

The module probably isn't readable/executeable from Asterisk

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, January 04, 2012 10:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)

 

Hi all,

I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently
my AGI is working fine in my two servers but not in my other four servers.
When  I tried execute an AGI (as a user asterisk) in command line it works
fine (even I also declare environmental variables in user profile and in my
AGI), but when I tried to call my AGI (perl) in dial plan, it don't get
executed.

Please advise me to resolve this issue.

-- 
Regards,

Ahmed Munir Chohan



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Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

how  can u give me a command?!..

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I tried also in asterisk 1.8 setting outbound variable  but didnt work
also 
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried  but still I get a video call

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 does not support setting the outbound codec.1.8 uses different
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source
for the 1.8 information,.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video
its sent to the other client as video call .I tried settings the
SIP_CODED_INBOUND and outbound also ... but no luck

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call
only cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its
like my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? Join us for a live introductory webinar every Thurs:
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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] Set Call Codec in extension.conf

[Danny Nicholas]

Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome;  1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

or 
exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

how  can u give me a command?!..

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I tried also in asterisk 1.8 setting outbound variable  but didnt work
also 
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried  but still I get a video call

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 does not support setting the outbound codec.1.8 uses different
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source
for the 1.8 information,.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video
its sent to the other client as video call .I tried settings the
SIP_CODED_INBOUND and outbound also ... but no luck

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call
only cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its
like my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,4,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,5,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,6,Queue(${EXTEN})

can any body help me with that?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to
Asterisk? 

Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

didnt work also :(

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:39 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome;  1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

or
exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

how  can u give me a command?!..

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I tried also in asterisk 1.8 setting outbound variable  but didnt work
also 
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried  but still I get a video call

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 does not support setting the outbound codec.1.8 uses different
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source
for the 1.8 information,.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video
its sent to the other client as video call .I tried settings the
SIP_CODED_INBOUND and outbound also ... but no luck

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call
only cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set Call Codec in extension.conf

Hi All,
I am trying to set call codec at extension.conf but it doesnt work ... its
like my command doesnt change anything


exten=6500,1,Answer
exten=6500,2,Playback(welcome)
exten=6500,3,SIPAddHeader(email:${SIP_HEADER(email)})

Re: [asterisk-users] Set Call Codec in extension.conf

[Danny Nicholas]

CLI output from call?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

didnt work also :(

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:39 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome;  1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

or
exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

how  can u give me a command?!..

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I tried also in asterisk 1.8 setting outbound variable  but didnt work
also 
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried  but still I get a video call

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:19 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 does not support setting the outbound codec.1.8 uses different
variables to set the outbound codec.  See UGRADE.txt in the Asterisk source
for the 1.8 information,.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:12 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

1.6 and 1.8  ... I tried changing stuff on both 
when I make audio call from my client which supports both audio and video
its sent to the other client as video call .I tried settings the
SIP_CODED_INBOUND and outbound also ... but no luck

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling
[ewiel...@nyigc.com]
Sent: Wednesday, January 04, 2012 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Providing which version of Asterisk you are using might be helpful.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

anyhelp guys?
 I tried a lot of stuff but it doesnt work  the Codec for audio call
only cannt be set...how I can set the call type video/audio at dail plan?

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 5:53 AM
To: 

Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-, SIP_CODEC=gsm
) in new stack
-- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-, SIP_CODEC_INB
OUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-, SIP_CODEC_OUT
BOUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in new 
stack
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin
g codec to 'gsm' for this call because of ${SIP_CODEC} variable
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin
g ${SIP_CODEC} variable because it is not shared by both ends.
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec: Changin
g codec to 'gsm' for this call because of ${SIP_CODEC} variable
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin
g ${SIP_CODEC} variable because it is not shared by both ends.
-- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-, welcome
) in new stack
[Jan  4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File welcome do
es not exist in any format
[Jan  4 17:50:16] WARNING[4131]: file.c:953 ast_streamfile: Unable to open welco
me (format 0x4 (ulaw)): No such file or directory
[Jan  4 17:50:16] WARNING[4131]: app_playback.c:471 playback_exec: ast_streamfil
e failed on SIP/6000- for welcome
-- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-, emai
l:fkha...@iconnecths.com) in new stack
-- Executing [6500@DLPN_DialPlan1:7] MixMonitor(SIP/6000-, 2012-0
1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b) in new stack
-- Executing [6500@DLPN_DialPlan1:8] Queue(SIP/6000-, 6500) in n
ew stack
-- Started music on hold, class 'default', on SIP/6000-
  == Begin MixMonitor Recording SIP/6000-

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

CLI output from call?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

didnt work also :(

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:39 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome;  1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

or
exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

how  can u give me a command?!..

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

My guess is that you should set the codec either before SIPADDHEADER or
before ANSWER.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I tried also in asterisk 1.8 setting outbound variable  but didnt work
also 
https://wiki.asterisk.org/wiki/display/AST/chan_sip+Channel+Variables
check the above ... I changed it and tried  

Re: [asterisk-users] Set Call Codec in extension.conf

[Danny Nicholas]

You are fighting a losing battle - you can't control the other end
Ignoring ${SIP_CODEC} variable because it is not shared by both ends.

You can probably do a SIP SET DEBUG ON and see what codecs are available on
the other end.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-,
SIP_CODEC=gsm
) in new stack
-- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-,
SIP_CODEC_INB
OUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-,
SIP_CODEC_OUT
BOUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in
new stack [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180
try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of
${SIP_CODEC} variable [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185
try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not
shared by both ends.
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec:
Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan
4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g
${SIP_CODEC} variable because it is not shared by both ends.
-- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-,
welcome
) in new stack
[Jan  4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File
welcome do es not exist in any format [Jan  4 17:50:16] WARNING[4131]:
file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No
such file or directory [Jan  4 17:50:16] WARNING[4131]: app_playback.c:471
playback_exec: ast_streamfil e failed on SIP/6000- for welcome
-- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-,
emai
l:fkha...@iconnecths.com) in new stack
-- Executing [6500@DLPN_DialPlan1:7] MixMonitor(SIP/6000-,
2012-0
1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b) in new
stack
-- Executing [6500@DLPN_DialPlan1:8] Queue(SIP/6000-, 6500)
in n ew stack
-- Started music on hold, class 'default', on SIP/6000-
  == Begin MixMonitor Recording SIP/6000-

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

CLI output from call?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

didnt work also :(

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:39 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome;  1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

or
exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:32 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

how  can u give me a command?!..

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:29 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

My guess is that you should set the codec either before 

Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

I am the other end  most codecs are available 
now my problem is when I make audio call using one side its converted to video 
call request (since my other end has also all codecs)
my app clients can do Audio and Video call,
now the Video call is ok
but the Audio part get converted to video request ...so I am trying to limit 
the codec to only audio codec...

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

You are fighting a losing battle - you can't control the other end
Ignoring ${SIP_CODEC} variable because it is not shared by both ends.

You can probably do a SIP SET DEBUG ON and see what codecs are available on
the other end.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-,
SIP_CODEC=gsm
) in new stack
-- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-,
SIP_CODEC_INB
OUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-,
SIP_CODEC_OUT
BOUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in
new stack [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180
try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of
${SIP_CODEC} variable [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185
try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not
shared by both ends.
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec:
Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan
4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g
${SIP_CODEC} variable because it is not shared by both ends.
-- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-,
welcome
) in new stack
[Jan  4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File
welcome do es not exist in any format [Jan  4 17:50:16] WARNING[4131]:
file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No
such file or directory [Jan  4 17:50:16] WARNING[4131]: app_playback.c:471
playback_exec: ast_streamfil e failed on SIP/6000- for welcome
-- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-,
emai
l:fkha...@iconnecths.com) in new stack
-- Executing [6500@DLPN_DialPlan1:7] MixMonitor(SIP/6000-,
2012-0
1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b) in new
stack
-- Executing [6500@DLPN_DialPlan1:8] Queue(SIP/6000-, 6500)
in n ew stack
-- Started music on hold, class 'default', on SIP/6000-
  == Begin MixMonitor Recording SIP/6000-

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

CLI output from call?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

didnt work also :(

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:39 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome;  1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

or
exten=6500,1,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)
exten=6500,n,Queue(${EXTEN})

-Original Message-
From: 

Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

Any suggestion will be great 

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib 
[fkha...@iconnecths.com]
Sent: Wednesday, January 04, 2012 11:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I am the other end  most codecs are available 
now my problem is when I make audio call using one side its converted to video 
call request (since my other end has also all codecs)
my app clients can do Audio and Video call,
now the Video call is ok
but the Audio part get converted to video request ...so I am trying to limit 
the codec to only audio codec...


--
_
-- 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

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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Set Call Codec in extension.conf

[Danny Nicholas]

Please post the sip.conf entries for 6000 and 6500.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I am the other end  most codecs are available 
now my problem is when I make audio call using one side its converted to
video call request (since my other end has also all codecs) my app clients
can do Audio and Video call, now the Video call is ok but the Audio part get
converted to video request ...so I am trying to limit the codec to only
audio codec...

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

You are fighting a losing battle - you can't control the other end Ignoring
${SIP_CODEC} variable because it is not shared by both ends.

You can probably do a SIP SET DEBUG ON and see what codecs are available on
the other end.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-,
SIP_CODEC=gsm
) in new stack
-- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-,
SIP_CODEC_INB
OUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-,
SIP_CODEC_OUT
BOUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in
new stack [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180
try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of
${SIP_CODEC} variable [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185
try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not
shared by both ends.
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec:
Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan
4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g
${SIP_CODEC} variable because it is not shared by both ends.
-- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-,
welcome
) in new stack
[Jan  4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File
welcome do es not exist in any format [Jan  4 17:50:16] WARNING[4131]:
file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No
such file or directory [Jan  4 17:50:16] WARNING[4131]: app_playback.c:471
playback_exec: ast_streamfil e failed on SIP/6000- for welcome
-- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-,
emai
l:fkha...@iconnecths.com) in new stack
-- Executing [6500@DLPN_DialPlan1:7] MixMonitor(SIP/6000-,
2012-0
1-05//2012-01-05_05:50:16_thursday_fkha...@iconnecths.com.wav,b) in new
stack
-- Executing [6500@DLPN_DialPlan1:8] Queue(SIP/6000-, 6500)
in n ew stack
-- Started music on hold, class 'default', on SIP/6000-
  == Begin MixMonitor Recording SIP/6000-

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:45 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

CLI output from call?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

didnt work also :(

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:39 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Move line 5 up to line 3 or line 1 (P.S. - the 1-6 numbering scheme is
cumbersome;  1-n-n-n-n-n is more practical).
Like this
exten=6500,1,Answer
exten=6500,n,Playback(welcome)
exten=6500,n,set(SIP_CODEC=gsm) -- this is not changed 
exten=6500,n,SIPAddHeader(email:${SIP_HEADER(email)})
exten=6500,n,MixMonitor(${STRFTIME(${EPOCH},GMT-6,%C%y-%m-%d//%C%y-%m-%d_%H:
%M:%S_%A)}_${SIP_HEADER(email)}.wav,b)

Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

there is nothing in sip.conf about what u asked
but 6500 is a queue with following info
[6500]
fullname = testing
strategy = rrmemory
timeout = 15
wrapuptime = 15
autofill = no
autopause = no
joinempty = yes
leavewhenempty = no
reportholdtime = no
maxlen = 0
musicclass = test
member = SIP/6251
member = SIP/6252
member = SIP/6253
member = SIP/6254

now the user 6251 is a user with following info and caller 6000

[6000]
username = 6000
transfer = yes
mailbox = 6000
call-limit = 100
type = peer
fullname = 6000
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6000
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = yes
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
disallow = all
allow = ulaw,gsm,h263,h263p,h264
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1
callcounter = yes

[6251]
username = 6251
transfer = yes
mailbox = 6251
call-limit = 100
type = peer
fullname = 6251
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6251
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = yes
hassip = yes
hasiax = yes
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
disallow = all
allow = ulaw,gsm,h263,h263p,h264
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1


From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Please post the sip.conf entries for 6000 and 6500.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I am the other end  most codecs are available 
now my problem is when I make audio call using one side its converted to
video call request (since my other end has also all codecs) my app clients
can do Audio and Video call, now the Video call is ok but the Audio part get
converted to video request ...so I am trying to limit the codec to only
audio codec...

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

You are fighting a losing battle - you can't control the other end Ignoring
${SIP_CODEC} variable because it is not shared by both ends.

You can probably do a SIP SET DEBUG ON and see what codecs are available on
the other end.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-,
SIP_CODEC=gsm
) in new stack
-- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-,
SIP_CODEC_INB
OUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-,
SIP_CODEC_OUT
BOUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in
new stack [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180
try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of
${SIP_CODEC} variable [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185
try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not
shared by both ends.
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec:
Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan
4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g
${SIP_CODEC} variable because it is not shared by both ends.
-- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-,
welcome
) in new stack
[Jan  4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File
welcome do es not exist in any format [Jan  4 17:50:16] WARNING[4131]:
file.c:953 ast_streamfile: Unable to open welco me (format 0x4 (ulaw)): No
such file or directory [Jan  4 17:50:16] WARNING[4131]: app_playback.c:471
playback_exec: ast_streamfil e failed on SIP/6000- for welcome
-- Executing [6500@DLPN_DialPlan1:6] SIPAddHeader(SIP/6000-,
emai
l:fkha...@iconnecths.com) in 

Re: [asterisk-users] Asterisk won't start - trap invalid opcode

[A J Stiles]

On Wednesday 04 January 2012, Duncan Turnbull wrote:
 I loaded the latest 1.6 which gets slightly further and a core dump shows
 this, but its past my ability to interpret
 
 # gdb -se asterisk -c core | tee /tmp/backtrace.txt
 GNU gdb (Ubuntu/Linaro 7.3-0ubuntu2) 7.3-2011.08
 Copyright (C) 2011 Free Software Foundation, Inc.
 License GPLv3+: GNU GPL version 3 or later
 http://gnu.org/licenses/gpl.html This is free software: you are free to
 change and redistribute it. There is NO WARRANTY, to the extent permitted
 by law.  Type show copying and show warranty for details.
 This GDB was configured as x86_64-linux-gnu.
 For bug reporting instructions, please see:
 http://bugs.launchpad.net/gdb-linaro/...
 Reading symbols from /usr/sbin/asterisk...done.
 [New LWP 19322]
 [New LWP 19323]
 [New LWP 19324]
 [New LWP 19325]
 [New LWP 19326]
 
 warning: Can't read pathname for load map: Input/output error.
 [Thread debugging using libthread_db enabled]
 Core was generated by `asterisk -d -g -cvvv'.
 Program terminated with signal 4, Illegal instruction.
 #0  0x00500eab in tzload (name=optimized out, sp=0x1fc7950,
 doextend=1) at stdtime/localtime.c:424 424static int tzload(const char
 *name, struct state * const sp, const int doextend)

It's a bit beyond my depth too, but I'd start with a look at localtime.c in 
the Asterisk source tree.  It might simply be trying to include something that 
isn't present on your system.

If you stick a /* harmless comment */ in this file and re-save it, this will 
give the file a new modification time.  Then run `make` again.  It will 
recompile just localtime.c  (this being the only source file that has changed 
since the last time make was run)  -- now watch very closely for errors.


Answers come *after* questions.

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Re: [asterisk-users] Rami

[Arjan Kroon | Mobillion]

Is this freeware, or a module which you can include in your ruby code?
Or is it a complete framework?


On 04 Jan 2012, at 5:31 PM, gokulnath wrote:

Hey,
There is a new kid in town if you want to code in ruby. Use 
adhearsionhttps://github.com/adhearsion/adhearsion/wiki, it's a framework to 
make voice apps.

On Wed, Jan 4, 2012 at 2:49 PM, Arjan Kroon | Mobillion 
arjan.kr...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote:
Hi,

Does anybody know if RAMI (Ruby Ami) is still functional?
And is this still compatible with asterisk 1.8

Best Regards,


Arjan Kroon
Mobillion BV

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Thanks  Regards
Gokulnath
@8129845320
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Re: [asterisk-users] Set Call Codec in extension.conf

[Danny Nicholas]

What about the allow/disallow lines in sip.conf?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

there is nothing in sip.conf about what u asked but 6500 is a queue with
following info [6500] fullname = testing strategy = rrmemory timeout = 15
wrapuptime = 15 autofill = no autopause = no joinempty = yes leavewhenempty
= no reportholdtime = no maxlen = 0 musicclass = test member = SIP/6251
member = SIP/6252 member = SIP/6253 member = SIP/6254

now the user 6251 is a user with following info and caller 6000

[6000]
username = 6000
transfer = yes
mailbox = 6000
call-limit = 100
type = peer
fullname = 6000
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6000
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = no
hassip = yes
hasiax = yes
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
disallow = all
allow = ulaw,gsm,h263,h263p,h264
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1
callcounter = yes

[6251]
username = 6251
transfer = yes
mailbox = 6251
call-limit = 100
type = peer
fullname = 6251
registersip = no
host = dynamic
callgroup = 1
type = peer
context = DLPN_DialPlan1
cid_number = 6251
hasvoicemail = no
vmsecret =
email =
threewaycalling = no
hasdirectory = no
callwaiting = no
hasmanager = no
hasagent = yes
hassip = yes
hasiax = yes
nat = yes
canreinvite = no
dtmfmode = rfc2833
insecure = no
pickupgroup = 1
disallow = all
allow = ulaw,gsm,h263,h263p,h264
autoprov = no
label =
macaddress =
linenumber = 1
LINEKEYS = 1


From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Please post the sip.conf entries for 6000 and 6500.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

I am the other end  most codecs are available 
now my problem is when I make audio call using one side its converted to
video call request (since my other end has also all codecs) my app clients
can do Audio and Video call, now the Video call is ok but the Audio part get
converted to video request ...so I am trying to limit the codec to only
audio codec...

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 11:54 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

You are fighting a losing battle - you can't control the other end Ignoring
${SIP_CODEC} variable because it is not shared by both ends.

You can probably do a SIP SET DEBUG ON and see what codecs are available on
the other end.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 11:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

-- Executing [6500@DLPN_DialPlan1:1] Set(SIP/6000-,
SIP_CODEC=gsm
) in new stack
-- Executing [6500@DLPN_DialPlan1:2] Set(SIP/6000-,
SIP_CODEC_INB
OUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:3] Set(SIP/6000-,
SIP_CODEC_OUT
BOUND=gsm) in new stack
-- Executing [6500@DLPN_DialPlan1:4] Answer(SIP/6000-, ) in
new stack [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180
try_suggested_sip_codec: Changin g codec to 'gsm' for this call because of
${SIP_CODEC} variable [Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6185
try_suggested_sip_codec: Ignorin g ${SIP_CODEC} variable because it is not
shared by both ends.
[Jan  4 17:50:16] NOTICE[4131]: chan_sip.c:6180 try_suggested_sip_codec:
Changin g codec to 'gsm' for this call because of ${SIP_CODEC} variable [Jan
4 17:50:16] NOTICE[4131]: chan_sip.c:6185 try_suggested_sip_codec: Ignorin g
${SIP_CODEC} variable because it is not shared by both ends.
-- Executing [6500@DLPN_DialPlan1:5] Playback(SIP/6000-,
welcome
) in new stack
[Jan  4 17:50:16] WARNING[4131]: file.c:650 ast_openstream_full: File
welcome do es not exist in any format 

Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

allow=all

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

What about the allow/disallow lines in sip.conf?

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Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)

[Ahmed Munir]

Hi,

I installed the modules in asterisk user home directory with read and
excitable permissions for asterisk but still my AGI not working.

Please provide me other advise to resolve this issue.


 Date: Wed, 4 Jan 2012 11:30:34 -0600
 From: Danny Nicholas da...@debsinc.com
 Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus
(oracle)
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
 Message-ID: 00ca01cccb06$911e8300$b35b8900$@debsinc.com
 Content-Type: text/plain; charset=us-ascii

 The module probably isn't readable/executeable from Asterisk



 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
 Sent: Wednesday, January 04, 2012 10:45 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)



 Hi all,

 I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
 Currently
 my AGI is working fine in my two servers but not in my other four servers.
 When  I tried execute an AGI (as a user asterisk) in command line it works
 fine (even I also declare environmental variables in user profile and in my
 AGI), but when I tried to call my AGI (perl) in dial plan, it don't get
 executed.

 Please advise me to resolve this issue.

 --
 Regards,

 Ahmed Munir Chohan


 -
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Set Call Codec in extension.conf

[Danny Nicholas]

Both sides?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

allow=all

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

What about the allow/disallow lines in sip.conf?

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Re: [asterisk-users] Set Call Codec in extension.conf

[Faraj Khasib]

yup  and video support is yes

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas 
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

Both sides?

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Faraj Khasib
Sent: Wednesday, January 04, 2012 12:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

allow=all

From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
[da...@debsinc.com]
Sent: Wednesday, January 04, 2012 12:09 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Set Call Codec in extension.conf

What about the allow/disallow lines in sip.conf?

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Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)

[Danny Nicholas]

What are the permissions on the module you are trying to run? (ls -l
/var/lib/asterisk/agi-bin/module)

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, January 04, 2012 12:15 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)

 

Hi,

I installed the modules in asterisk user home directory with read and
excitable permissions for asterisk but still my AGI not working.

Please provide me other advise to resolve this issue.
 

Date: Wed, 4 Jan 2012 11:30:34 -0600
From: Danny Nicholas da...@debsinc.com
Subject: Re: [asterisk-users] asterisk - AGI (perl) - sqlplus
   (oracle)
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
   asterisk-users@lists.digium.com
Message-ID: 00ca01cccb06$911e8300$b35b8900$@debsinc.com
Content-Type: text/plain; charset=us-ascii

The module probably isn't readable/executeable from Asterisk



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, January 04, 2012 10:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)



Hi all,

I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently
my AGI is working fine in my two servers but not in my other four servers.
When  I tried execute an AGI (as a user asterisk) in command line it works
fine (even I also declare environmental variables in user profile and in my
AGI), but when I tried to call my AGI (perl) in dial plan, it don't get
executed.

Please advise me to resolve this issue.

--
Regards,

Ahmed Munir Chohan



- 
Regards,

Ahmed Munir Chohan



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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Bruce B]

 Note to self: Never release anything asterisk related without testing
 on RHEL/Centos 5

 Thank you for reporting this. I have replaced sox with flac and it seems
 to work now on older platforms too (tested on Centos 5 with asterisk 1.4).
 You can get the updated code here:
 https://github.com/zaf/asterisk-speech-recog/tarball/master

 
 Lefteris Zafiris


Works beautifully. Amazing job Lefteris. Thanks.

The best result I got in probability was 0.9725632 by saying, hello. I
think there is some non-phonetic logic built-in as well. I tried, 1, 2
and I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got,
0.97256315. Probably Google sees the pattern?!

What are some of the other tricks (if any) or consideration that one should
make while creating a strong speech recognition enabled IVR?

Best,
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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[isrlgb]

Does anyone know what languages are supported?
-Original Message-
From: Bruce B bruceb...@gmail.com
Sender: asterisk-users-boun...@lists.digium.com
Date: Wed, 4 Jan 2012 13:25:18 
To: Asterisk Users Mailing List - Non-Commercial 
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Speech recognition in asterisk using google
 voice API

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Michelle Dupuis]

Wow - nice!  A few quick questions:

1.  How long can the recording be for translation?
2.  Any limitation on how much text the return (transcribed) variable can hold?
3.  Any commercial / terms of use limitations?

From: asterisk-users-boun...@lists.digium.com 
[asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B 
[bruceb...@gmail.com]
Sent: Wednesday, January 04, 2012 1:25 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Speech recognition in asterisk using google voice 
API

Note to self: Never release anything asterisk related without testing
on RHEL/Centos 5

Thank you for reporting this. I have replaced sox with flac and it seems
to work now on older platforms too (tested on Centos 5 with asterisk 1.4).
You can get the updated code here:
https://github.com/zaf/asterisk-speech-recog/tarball/master


Lefteris Zafiris


Works beautifully. Amazing job Lefteris. Thanks.

The best result I got in probability was 0.9725632 by saying, hello. I think 
there is some non-phonetic logic built-in as well. I tried, 1, 2 and I got 
0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got, 0.97256315. 
Probably Google sees the pattern?!

What are some of the other tricks (if any) or consideration that one should 
make while creating a strong speech recognition enabled IVR?

Best,

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Re: [asterisk-users] Rami

[Steve Edwards]

On Wed, 4 Jan 2012, Arjan Kroon | Mobillion wrote:

Is this freeware, or a module which you can include in your ruby code?Or 
is it a complete framework?


Is this list faster than Google?

--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Asterisk won't start - trap invalid opcode

[Steve Edwards]

On Wed, 4 Jan 2012, A J Stiles wrote:


If you stick a /* harmless comment */ in this file and re-save it, this will
give the file a new modification time.  Then run `make` again.  It will
recompile just localtime.c  (this being the only source file that has changed
since the last time make was run)  -- now watch very closely for errors.


The 'touch' command will update the file's access and modification times* 
without the risk of trashing something in the file.


*) Command line parameters can select just the access or the modification 
time. The default is both.


--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Lefteris Zafiris]

On Wed, Jan 4, 2012 at 8:47 PM, Michelle Dupuis mdup...@ocg.ca wrote:
 Wow - nice!  A few quick questions:

 1.  How long can the recording be for translation?
At the moment the recording timeout is set at 15sec. I haven't tested
yet the max
length  of voice data ta google accepts (all this voice recognition
stuff is undocumented).
I have read that it is between 10-20 seconds but havent really went to
test this yet. On my todo list is
to add the option to cut the sound data in smaller chunks before
sending them to google and get rid of the
recording length limitations.

 2.  Any limitation on how much text the return (transcribed) variable can
 hold?
This better be answered by the astsrisk devs but empirically talking i
have loaded in dialplan variables really big
chunks of text (like the complete gpl license) without having any problems.

 3.  Any commercial / terms of use limitations?
This is a gray area at the moment. Voice recognition is undocumented
in google's API and i guess not
officially supported yet. I hope it gets covered by the general TOS of
google services:
 http://www.google.com/accounts/TOS


Lefteris Zafiris

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Lefteris Zafiris]

On Wed, Jan 4, 2012 at 8:27 PM,  isr...@gmail.com wrote:
 Does anyone know what languages are supported?

For sure english and spanish, since its undocumented i don't have a
complete list
yet.


Lefteris Zafiris

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Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)

[Steve Edwards]

Un-top-posting...

On Wed, 4 Jan 2012, Ahmed Munir wrote:

I'm trying to run an AGI in PERL which uses the module DBD-Oracle. 
Currently my AGI is working fine in my two servers but not in my other 
four servers. When I tried execute an AGI (as a user asterisk) in 
command line it works fine (even I also declare environmental variables 
in user profile and in my AGI), but when I tried to call my AGI (perl) 
in dial plan, it don't get executed.


It usually boils down to PATHs, environment variables, permissions, 
missing script interpreters, etc.


When you execute your AGI from the command line, are you passing a valid 
AGI environment to the AGI via STDIN? If not, you may be violating the AGI 
protocol, which, is probably not why your 'AGI' is not executing, but will 
bite you some time in the future.


When you say 'it don't get executed' do you mean that Asterisk cannot 
locate the AGI at all or do you mean it does not execute as you expect?


On Wed, 4 Jan 2012, Ahmed Munir wrote:

I installed the modules in asterisk user home directory with read and 
excitable permissions for asterisk but still my AGI not working.


Asterisk looks for AGI executables in the 'astagidir' (AKA ASTAGIDIR) 
directory which is usually /var/lib/asterisk/agi-bin/. This can be set in 
asterisk.conf. The path to this file can be specified on the command line 
used to start asterisk.


It would be unusual for ASTAGIDIR to be set the asterisk user's home 
directory. It would also be unusual for the asterisk user's home directory 
to be set to ASTAGIDIR.


--
Thanks in advance,
-
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Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Lefteris Zafiris]

 Works beautifully. Amazing job Lefteris. Thanks.

 The best result I got in probability was 0.9725632 by saying, hello. I
 think there is some non-phonetic logic built-in as well. I tried, 1, 2 and
 I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got,
 0.97256315. Probably Google sees the pattern?!

 What are some of the other tricks (if any) or consideration that one should
 make while creating a strong speech recognition enabled IVR?

Google accepts sound files at any sampling rate (up to 44.1kHz) so if
you can use some wideband codec ( eg g722)
It can greatly improve the sound quality and the detection rates. For
now the script supports 8kHz and 16kHz sampling rates
for recording and it can be set by editing the scripts user defined
parameters ( the variable $samplerate).
Anything that improves the recording sound clarity will help, a good
phone, low background noise level etc.
I have also read that normalizing the recording and setting the gain
to -5 db improves detection rates. I m experimenting with this at the
moment and there will be some new code soon (as soon as i get sox
working in RHEL/Centos 5 :P ).


Lefteris Zafiris

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[asterisk-users] From address missing 'sip:', using it anyway

[motty.cruz]

Hello, 
I see the following error in the logs

[Jan  4 11:37:35] NOTICE[21]: chan_sip.c:15388 check_user_full: From address
missing 'sip:', using it anyway

Does anybody know how to stop this error? It does not seem to be affecting
performance on the Asterisk 1.8.4.1 running on Centos Linux 2.6. I have
google it but empty! 

Thanks, 
Celso 


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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[sean darcy]

On 1/4/2012 2:26 PM, Lefteris Zafiris wrote:


Works beautifully. Amazing job Lefteris. Thanks.

The best result I got in probability was 0.9725632 by saying, hello. I
think there is some non-phonetic logic built-in as well. I tried, 1, 2 and
I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got,
0.97256315. Probably Google sees the pattern?!

What are some of the other tricks (if any) or consideration that one should
make while creating a strong speech recognition enabled IVR?


Google accepts sound files at any sampling rate (up to 44.1kHz) so if
you can use some wideband codec ( eg g722)
It can greatly improve the sound quality and the detection rates. For
now the script supports 8kHz and 16kHz sampling rates
for recording and it can be set by editing the scripts user defined
parameters ( the variable $samplerate).
Anything that improves the recording sound clarity will help, a good
phone, low background noise level etc.
I have also read that normalizing the recording and setting the gain
to -5 db improves detection rates. I m experimenting with this at the
moment and there will be some new code soon (as soon as i get sox
working in RHEL/Centos 5 :P ).



This is really spectacular. Thanks.

I'm running Fedora 15, so I can use flac or sox. Any reason to prefer 
one over the other?


sean



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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Israel Gottlieb]

wow i just tried in hebrew and i'll say just 1 word WOW

On Wed, Jan 4, 2012 at 9:48 PM, sean darcy seandar...@gmail.com wrote:

 On 1/4/2012 2:26 PM, Lefteris Zafiris wrote:


 Works beautifully. Amazing job Lefteris. Thanks.

 The best result I got in probability was 0.9725632 by saying, hello. I
 think there is some non-phonetic logic built-in as well. I tried, 1, 2
 and
 I got 0.86534226 in accuracy. While I tried 1, 2, 3, 4, 5 I got,
 0.97256315. Probably Google sees the pattern?!

 What are some of the other tricks (if any) or consideration that one
 should
 make while creating a strong speech recognition enabled IVR?


 Google accepts sound files at any sampling rate (up to 44.1kHz) so if
 you can use some wideband codec ( eg g722)
 It can greatly improve the sound quality and the detection rates. For
 now the script supports 8kHz and 16kHz sampling rates
 for recording and it can be set by editing the scripts user defined
 parameters ( the variable $samplerate).
 Anything that improves the recording sound clarity will help, a good
 phone, low background noise level etc.
 I have also read that normalizing the recording and setting the gain
 to -5 db improves detection rates. I m experimenting with this at the
 moment and there will be some new code soon (as soon as i get sox
 working in RHEL/Centos 5 :P ).


 This is really spectacular. Thanks.

 I'm running Fedora 15, so I can use flac or sox. Any reason to prefer one
 over the other?

 sean




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Re: [asterisk-users] Failed to authenticate on INVITE to Anonymous

[sean darcy]

On 1/4/2012 4:37 AM, Jayesh Labade wrote:

Please help me..

Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com



On Wed, Jan 4, 2012 at 12:51 PM, Jayesh Labade jayesh.lab...@gmail.com
mailto:jayesh.lab...@gmail.com wrote:

Hello Experts,

I have pasted my issue in http://pastebin.com/zBGVmdcY

I Cant able to Originate call from SIp trunk..I got this [Jan 3
11:52:08] NOTICE[29823]: chan_sip.c:19718 handle_response_invite:
Failed to authenticate on INVITE to 'Anonymous
sip:test02@anonymous.invalid;tag=as57d3a806'
i am unable to make outbound call from this trunk. while if i
registered this trunk in softphone like Xlite, there is no problem
with outbound calls. Help me.

please find sip.conf file in http://pastebin.com/zBGVmdcY

I have pasted sip debug with verbosity of failed call
http://pastebin.com/jL2ki0s8


Best Regards,
*Jayesh Labade*
e-mail: jayesh.lab...@gmail.com mailto:jayesh.lab...@gmail.com




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Try:
register = test02:test02@192.168.1.55/s

sean



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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Lefteris Zafiris]

On Wed, 04 Jan 2012 14:48:22 -0500
sean darcy seandar...@gmail.com wrote:

 This is really spectacular. Thanks.
 
 I'm running Fedora 15, so I can use flac or sox. Any reason to prefer 
 one over the other?
 
 sean

We have to convert the voice data to flac format before sending them to
google, this can be done by both sox and flac encoder. For now the
script uses flac encoder for compatibility with older distros (mainly
RHEL 5). Sox is a bit more flexible and also gives you the option to
edit the sound data (normalizing, changing levels etc).


Lefteris Zafiris

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Re: [asterisk-users] Asterisk won't start - trap invalid opcode

[Duncan Turnbull]

On 5/01/2012, at 8:06 AM, Steve Edwards wrote:

 On Wed, 4 Jan 2012, A J Stiles wrote:
 
 If you stick a /* harmless comment */ in this file and re-save it, this will
 give the file a new modification time.  Then run `make` again.  It will
 recompile just localtime.c  (this being the only source file that has changed
 since the last time make was run)  -- now watch very closely for errors.
 
 The 'touch' command will update the file's access and modification times* 
 without the risk of trashing something in the file.
 
 *) Command line parameters can select just the access or the modification 
 time. The default is both.
 

Touch seemed safer but I didn't see any errors

:/usr/src/asterisk-1.6.2.22# make
   [CC] stdtime/localtime.c - stdtime/localtime.o
   [LD] abstract_jb.o acl.o adsistub.o aescrypt.o aeskey.o aestab.o alaw.o 
app.o ast_expr2.o ast_expr2f.o asterisk.o astfd.o astmm.o astobj2.o audiohook.o 
autoservice.o bridging.o callerid.o cdr.o channel.o chanvars.o cli.o config.o 
cryptostub.o datastore.o db.o devicestate.o dial.o dns.o dnsmgr.o dsp.o enum.o 
event.o features.o file.o fixedjitterbuf.o frame.o fskmodem.o 
global_datastores.o hashtab.o heap.o http.o image.o indications.o io.o 
jitterbuf.o loader.o logger.o manager.o md5.o netsock.o pbx.o plc.o poll.o 
privacy.o rtp.o say.o sched.o sha1.o slinfactory.o srv.o ssl.o 
stdtime/localtime.o strcompat.o strings.o taskprocessor.o tcptls.o tdd.o term.o 
test.o threadstorage.o timing.o translate.o udptl.o ulaw.o utils.o version.o 
xml.o xmldoc.o editline/libedit.a db1-ast/libdb1.a - asterisk
 +- Asterisk Build Complete -+
 + Asterisk has successfully been built, and +
 + can be installed by running:  +
 +   +
 +make install   +
 +---+

I haven't found anything obvious in the debug stuff although I am not familiar 
enough to be sure 

Thanks very much

Unless there is something obvious I am thinking I will revert to either an 
earlier OS or maybe 32 bit - although that seems excessive

Cheers Duncan

 -- 
 Thanks in advance,
 -
 Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
 Newline  Fax: +1-760-731-3000
 
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[asterisk-users] question sangoma vs digium

[Agustina Berretta]

Hi!
Hello! I wanted to know if you have experienced problems installing both a
Sangoma and a Digium card in the same Server.

Thnks a lot!
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[asterisk-users] DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released

[Asterisk Development Team]

The Asterisk Development Team is pleased to announce the first
release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0.

2.6.0 is a feature release which:

  - Adds support for the TE820 8-span card to the wct4xxp driver.

  - Decrease load time of analog cards supported by the wctdm24xxp
driver.

  - Adds sysfs object model to facilitate persistent span numbering
and early loading of modules (NOTE: by default this release
still behaves like previous releases with regards to span
numbering assignment).

  - dahdi_pcap tool is now included in DAHDI-tools but not compiled
by default since it depends on a currently unsupported interface
in DAHDI-Linux. It is intended that in future releases this will
be compiled by default.

Issues closed in this release:

DAHTOOL-49: adding pcap support to Dahdi
(Reported by: Torrey Searle)
DAHLIN-258: weird sound with a native bridged isdn-bri connection
(Reported by: Daniel)
DAHLIN-264: xpp: E1 CAS multiframe bits not properly set

DAHDI-Linux 2.6.0, DAHDI-Tools 2.6.0, and DAHDI-Linux-Complete
2.6.0+2.6.0 are available for immediate download at:

http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

The DAHDI-Linux shortlog of changes that are not in 2.5.0.2:

 Doug Bailey:
wctdm24xxp, wcte12xp: Update VPMOCT032 firmware to 1.12.0.
 
 Tzafrir Cohen:
Avoid building PCI devices if kernel has no PCI
xpp: Allow up to 128 Astribanks on a system
xpp: increase command queue length to 1500
xpp: USB_FW rev 10085: fix regression from r10013
xpp: PIC_TYPE_1 rev 9841: followup to r10013
bugfix: off-by-one in span assignment
xpp: USB firmware r9964: minor bugfixes
xpp: bugfix: clear NOTOPEN span alarm on assign
xpp: bugfix -- manage xpd refcount for EC module
xpp: Adaptations for E-Main-3
xpp: remove leftovers of old XPD_STATE method
README: Minor additions regarding pinned-spans
README: initial update for span assignments
dahdi: Add error messages in dahdi_ioctl_chanconfig.
xpp: fix FXS D DTMF detection (not zero)
xpp: fix bashism in xpp_debug
live_dahdi: optionally generate FreePBX DB entries
 
 Matthew Fredrickson:
wct4xxp: Add support for TE820 and VPMOCT256.
 
 Russ Meyerriecks:
wct4xxp: Remove vpm400 support.
wct4xxp: Revise vpm struct due to product name changes
wct4xxp: Handle incorrect vpm module/card pairings
wct4xxp: minor: Removed unnecessary instrumentation
wct4xxp: Expose serial number in dahdi_device and kernel log.
wct4xxp: Add field upgradable firmware support for TE820.
wcte12xp, wctdm24xxp: Remove frowny face from vpmoct032 error message
 
 Oron Peled:
xpp: BRI: batch D-Channel packets to fix frag.
xpp: BRI: split multibyte functionality
xpp: BRI: remove trivial BRISTUFF wrappers
xpp: BRI: remove legacy BRISTUFF code
xpp: bad module_put() when too many Astribanks
DAHDI-linux: Fix surprise removal problems
xpp: BRI: fix timing priority calculation
xpp: FXS: mwi and search_fsk fixes
xpp: PRI: restore pri_protocol to R/W:
xpp: pri: fix RS1 init in E1 CAS mode
xpp: fxs: demote SETPOLARITY message to DBG()
xpp: silence some bad ioctl() reporting
xpp: restore backward compat dahdi_registration
Extra debugging aids and messages
xpp: cleanup some printk()'s
added 'basechan' and 'channels' attributes to spans
dahdi: Give userspace a chance to respond to surprise removal.
xpp: Remove dahdi_autoreg parameter:
xpp: more informative span description:
xpp: make unregistration safer (idempotent)
xpp: adapt to 'location' attribute removal:
xpp: PRI: use DAHDI new set_spantype() method
dahdi: Expose spans in sysfs.
dahdi: dahdi_is_analog_span() - dahdi_is_digital_span()
dahdi: start handling surprise device removal.
 
 Shaun Ruffell:
wctdm24xxp: Fix bug if hook state on FXS changes before channel 
configuration.
wct4xxp: Reduce time spent waiting for auth done bit on TE820.
wct4xxp: Fail startup if not generating interrupts.
dahdi: Return dahdi_span_ops.startup callback errors to userspace.
wctdm24xxp: Do not call voicebus_release() before 
wctdm_back_out_gracefully()
dahdi: #include linux/module.h in dahdi/kernel.h and GpakCust.h
wctc4xxp: Replace 'ndo_set_multicast_list' with 'set_rx_mode'
wctdm24xxp: Wait for background threads to complete on failed load.
dahdi: Unregister dahdi_device from sysfs if we fail to auto assign 
spans.
dahdi: Fix typo in previous commit which forced some spans to always 
fail assignment.
dahdi: 

Re: [asterisk-users] asterisk 1.8 codec negotiation

[Kevin P. Fleming]

On 01/01/2012 04:17 PM, cov...@ccs.covici.com wrote:

Hi.  I am using asterisk 1.8 and everything was working fine when I was
at svn  342661.  I then upgraded to vrsion 349339 and discovered the
following problem -- one of the end points is a freeswitch box which
offers a number of codecs, including PCMU.  However, when I tried to
make a call I got a 488 response and  a message multiple audio streams
not supported in the log.


multiple audio streams != multiple audio codecs. For some reason 
Asterisk is receiving an INVITE with an offer for more than one audio 
stream (m=audio), and that is not supported.


--
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Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] NAT/IPTABLES workarounds

[Kevin P. Fleming]

On 01/03/2012 10:03 AM, Patrick Lists wrote:

On 03-01-12 16:24, Danny Nicholas wrote:

Hello List,

I work in an environment where I have to request IPTABLES changes rather
than doing them myself. Is there a way to utilize my SSH (port 22) to
get a functional (and with good sound) Asterisk installation with
multiple channels up without requesting the 5060(SIP) 5061 (TLS) and
UDP/RTP (usually 10001-2) IPTABLES allowances?


Not with SIP as it needs a port for signaling (usually 5060) and RTP
ports for sending the actual voice packets. So for SIP you will always
need multiple ports. If you can use IAX then you could use port 22 as
IAX only needs one port. The question is how are you going to SSH into
the box if you use the SSH port for Asterisk?


It is not practical (although not impossible) to run UDP over an SSH 
tunnel. Since VoIP media is generally transported over UDP, this will be 
a major obstacle.


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Re: [asterisk-users] Anyone have a reliable T.38 Solution

[Kevin P. Fleming]

On 01/04/2012 12:25 AM, Matt Darnell wrote:

Aloha,

We are looking to roll a solution that will have the following network layout:

ISDN-PRI--  Asterisk--  T.38--  ATA--  Fax

Does version 1.8 with the Digium fax driver have this capability?  I
like 1.8 because it is a long term support version.

What ATA's are people using?

Any working solutions would be great!


What you are looking for is T.38 gateway mode (converting between T.30 
over modems on a TDM circuit and T.38 over UDPTL), and the answer is no: 
Asterisk 1.8 does not have T.38 gateway mode. Asterisk 10 does, and it 
is supported using SpanDSP and res_fax_spandsp. It is not yet supported 
by Digium's Fax for Asterisk commercial FAX module.


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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Asterisk won't start - trap invalid opcode

[James Cloos]

 DT == Duncan Turnbull dun...@e-simple.co.nz writes:

DT I have a new install of asterisk 1.8.8.1 on ubuntu server 
DT 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64 
x86_64 GNU/Linux

DT The only errors I can see are limited - I also stopped wan router and dahdi 
and I still get 
DT ~# asterisk -cvv
DT Illegal instruction

What does /proc/cpuinfo say?  (Just the first chunk is enough.)

Try running asterisk is gdb:

:; gdb asterisk

(gdb) run -cvvddd

When it dies, try:

(gdb) bt full

(gdb) disasemble /m

You may also want to recompile asterisk after turing on:

DONT_OPTIMIZE
DEBUG_THREADS
BETTER_BACKTRACES

in the Compiler Flags section of make menuselect.

The gdb output if you do that may be more comprehensible.

Either way run gdb from the asterisk src directory.

When you find the point where it crashed, you can discover what the
illegal instruction is.

I suspect your compile may expect a more recent cpu than you have, and
may use sse instructions which it doesn't support.  A disassembly around
the failing instruction will confirm whether that is true and which
instruction it is.

-JimC
-- 
James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6

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Re: [asterisk-users] Speech recognition in asterisk using google voice API

[Lefteris Zafiris]

Fresh code is out! The use of sox can be now optionally enabled by the
user if the system has a recent version of the program (won't work in
RHEL/Centos 5)
This is done by editing the script and setting the variable 'use_sox'.
When sox is used the audio gets normalized, low frequency noise (100Hz)
is removed and also possible DC offset is corrected. Those are supposed
to improve the recognition results(?). The settings are still a bit
experimental, feel free to play with them and report what settings
improved your results.

get the new version here:
https://github.com/downloads/zaf/asterisk-speech-recog/asterisk-speech-recog-0.3.tar.gz


Lefteris Zafiris

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Re: [asterisk-users] Asterisk won't start - trap invalid opcode

[Duncan Turnbull]

On 5/01/2012, at 12:21 PM, James Cloos wrote:

 DT == Duncan Turnbull dun...@e-simple.co.nz writes:
 
 DT I have a new install of asterisk 1.8.8.1 on ubuntu server 
 DT 3.0.0-14-server #23-Ubuntu SMP Mon Nov 21 20:49:05 UTC 2011 x86_64 x86_64 
 x86_64 GNU/Linux
 
 DT The only errors I can see are limited - I also stopped wan router and 
 dahdi and I still get 
 DT ~# asterisk -cvv
 DT Illegal instruction
 
 What does /proc/cpuinfo say?  (Just the first chunk is enough.)
 
 Try running asterisk is gdb:
 
:; gdb asterisk
 
(gdb) run -cvvddd
 
 When it dies, try:
 
(gdb) bt full
 
(gdb) disasemble /m
 
 You may also want to recompile asterisk after turing on:
 
DONT_OPTIMIZE
DEBUG_THREADS
Hi James

I think the DONT_OPTIMIZE flag made a difference, the system is not crashing 
anymore

I am going to test it, and see if its really back, the other detail looked 
fairly similar to the core dump output in previous emails but there wasn't 
anything I could easily discern

I will let you all know how it turns out - thanks everyone

Cheers Duncan

BETTER_BACKTRACES
 
 in the Compiler Flags section of make menuselect.
 
 The gdb output if you do that may be more comprehensible.
 
 Either way run gdb from the asterisk src directory.
 
 When you find the point where it crashed, you can discover what the
 illegal instruction is.
 
 I suspect your compile may expect a more recent cpu than you have, and
 may use sse instructions which it doesn't support.  A disassembly around
 the failing instruction will confirm whether that is true and which
 instruction it is.
 
 -JimC
 -- 
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Re: [asterisk-users] asterisk 1.8 codec negotiation

[covici]

Kevin P. Fleming kpflem...@digium.com wrote:

 On 01/01/2012 04:17 PM, cov...@ccs.covici.com wrote:
  Hi.  I am using asterisk 1.8 and everything was working fine when I was
  at svn  342661.  I then upgraded to vrsion 349339 and discovered the
  following problem -- one of the end points is a freeswitch box which
  offers a number of codecs, including PCMU.  However, when I tried to
  make a call I got a 488 response and  a message multiple audio streams
  not supported in the log.
 
 multiple audio streams != multiple audio codecs. For some reason
 Asterisk is receiving an INVITE with an offer for more than one audio
 stream (m=audio), and that is not supported.
OK, but if I have a phone or in my case a server which offers a choice
of codecs, why can't asterisk just pick the ones it has rather than
reject the call?  Is there a way to do this correctly as far as asterisk
is concerned?


-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

 John Covici
 cov...@ccs.covici.com

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Re: [asterisk-users] question sangoma vs digium

[James zhu]

hello:
i think it can be done, please refer this link:
http://wiki.sangoma.com/Asterisk-FAQ#Digium
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com 


Date: Wed, 4 Jan 2012 18:47:28 -0200
From: agustina.berre...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] question sangoma vs digium

Hi!
Hello! I wanted to know if you have experienced problems installing both a 
Sangoma and a Digium card in the same Server.

Thnks a lot!


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[asterisk-users] 回覆︰ dialplan - dial command - custom ringtone

[Qqblog Qqblog]

my config:
hardphone - pstn gateway - asterisk - pstn gateway - hardphone

i am using asterisk 1.4.xx

w option is Dial is for recording. how does it related to ringtone?

pls advise.





 從︰ Carlos Rojas crt.ro...@gmail.com
收件人︰ Qqblog Qqblog qqb...@ymail.com; Asterisk Users Mailing List - 
Non-Commercial Discussion asterisk-users@lists.digium.com 
傳送日期︰ 2012年01月3日 (週二) 8:42 PM
主題︰ Re: [asterisk-users] dialplan - dial command - custom ringtone
 

Hello
Do you use hard phone or softphone?
In many ip phones you can change the ring tones or use w option in Dial 
command
Regards
On Jan 3, 2012 4:08 AM, Qqblog Qqblog qqb...@ymail.com wrote:

i could add r option in dial command. this will generate a ringtone during 
connection. could i change this default ringtone?


i tried indications.conf but not success.

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Re: [asterisk-users] Anyone have a reliable T.38 Solution

[Matt Darnell]

On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn
da...@klaverstyn.com.au wrote:
 I'm using  the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and 
 rx_fax on multiple installations with no problems.

David,

Are you running 10.0 or 1.8?

Glad to know that the PAP2T has a solid T.38 implementation!

-Matt

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Re: [asterisk-users] Anyone have a reliable T.38 Solution

[Klaverstyn, David C]

I'm using 1.8, but also have 1.4 and 1.2 installs using the same configuration.

Regards
David.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Darnell
Sent: Thursday, 5 January 2012 1:55 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Anyone have a reliable T.38 Solution

On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn da...@klaverstyn.com.au 
wrote:
 I'm using  the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and 
 rx_fax on multiple installations with no problems.

David,

Are you running 10.0 or 1.8?

Glad to know that the PAP2T has a solid T.38 implementation!

-Matt

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Re: [asterisk-users] Server-to-server BLF

[Ronald Cepres]

Hi Kevin,

Thanks for your suggestion.

On the website of OpenAIS, it seems that it is not supported anymore and
their download links (FTP and SVN) are broken (been trying it for about a
month now). Is it still possible to use OpenAIS method? The other solution
on the wiki is using XMPP which is for jabber. IMHO, it means that the XMPP
solution can't be used on SIP peers, right?

Regards,
Ronald

On Thu, Nov 17, 2011 at 1:01 AM, Kevin P. Fleming kpflem...@digium.comwrote:

 On 11/16/2011 04:18 AM, Ronald Cepres wrote:

 Hi all,

 Do you have an idea on the best way on how to implement a system with
 multiple Asterisk servers with BLF working in such a way that a peer on
 one server can subscribe to another peer on the other server in a
 seamless manner? Has anyone set-up a system like this before?


 Here is one way:

 https://wiki.asterisk.org/**wiki/display/AST/Distributed+**
 Device+State+with+AIShttps://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State+with+AIS

 There are other methods documented on the wiki as well.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Set Call type in dial plan

[Sammy Govind]

Hi,
Sorry for late reply. Hope you've already found out something about it.

What version of asterisk you are using, that function for choosing
inbound/outbound call leg codecs is for newer versions of asterisk.
See these pages:
http://www.voip-info.org/wiki/view/Asterisk+variables
https://issues.asterisk.org/view.php?id=13243

Regards,
Sammy


On Tue, Jan 3, 2012 at 2:31 PM, Faraj Khasib fkha...@iconnecths.com wrote:

 thats excatly what I want, can u plz give me the command, I want to choose
 only ulow
 
 From: asterisk-users-boun...@lists.digium.com [
 asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind [
 govoi...@gmail.com]
 Sent: Tuesday, January 03, 2012 3:26 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Set Call type in dial plan

 Hi,

 For such call you just need to select the outbound codec before the dial()
 app.

 choose the audio-only codecs and thus no video codec strings will be
 exchanged in that call.

 --
 Regards,
 Sammy

 On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib fkha...@iconnecths.com
 mailto:fkha...@iconnecths.com wrote:
 this is what my SIP Invite message when I make Video call

 INVITE sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102 SIP/2.0
 Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport
 From: sip:6097@192.168.21.102mailto:sip%3A6097@192.168.21.102
 ;tag=1857098215
 To: sip:6500@192.168.21.102mailto:sip%3A6500@192.168.21.102
 Contact: sip:6097@192.168.21.193:52933
 ;transport=udp;+g.oma.sip-im;language=en,fr;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
 Call-ID: b9453704-d76a-b8ce-3247-c999abff7395
 CSeq: 324677463 INVITE
 Content-Type: application/sdp
 Content-Length: 588
 Max-Forwards: 70
 Route: sip:192.168.21.102:5060;lr;transport=udp
 Accept-Contact:
 *;+g.3gpp.icsi-ref=urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel
 P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
 Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE,
 REFER
 Privacy: none
 P-Access-Network-Info: ADSL;utran-cell-id-3gpp=
 User-Agent: Medcor
 Supported: 100rel

 v=0
 o=doubango 1983 678901 IN IP4 192.168.21.193
 s=-
 c=IN IP4 192.168.21.193
 t=0 0
 m=audio 36372 RTP/AVP 8 0 9 101
 a=ptime:20
 a=rtpmap:8 PCMA/8000/1
 a=rtpmap:0 PCMU/8000/1
 a=rtpmap:9 G722/8000/1
 a=rtpmap:101 telephone-event/8000/1
 a=fmtp:101 0-15
 m=video 59296 RTP/AVP 125 106 121 103
 a=rtpmap:125 VP8/9
 a=fmtp:125 CIF=2;QCIF=2;SQCIF=2
 a=rtpmap:106 H264/9
 a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452;
 max-mbps=11880
 a=rtpmap:121 MP4V-ES/9
 a=fmtp:121 profile-level-id=3
 a=rtpmap:103 H263-1998/9
 a=fmtp:103 CIF=2;QCIF=2;SQCIF=2

 when I make Audio call requests I dont have the video part  but at
 receiver since two clients can make video call they have Asterisks adds the
 Video Part in request sent to receiver,I dont want that part added , how I
 can delete it ?
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Re: [asterisk-users] DIALSTATUS Values

[Kamlesh Kumar]

Can anybody please reply on this?
 
Regards,
Kamlesh
 



From: kamlesh_...@hotmail.com
To: asterisk-users@lists.digium.com
Date: Tue, 27 Dec 2011 09:49:21 +
Subject: Re: [asterisk-users] DIALSTATUS Values





Hello,
 
After investing some time, I could come to know the reason for not getting the 
data value is that if I use system command with any of asterisk cli command as 
given below, data value is returned blank.
 
$output=system(/usr/sbin/asterisk -rx 'sip show peers' | grep OK | cut -f 1 -d 
/ | grep '100' )
 
Could you please suggest now how to rectify this?
 
Regards,
Kamlesh
 


 To: asterisk-users@lists.digium.com
 From: t...@softins.co.uk
 Date: Fri, 2 Dec 2011 12:27:19 +
 Subject: Re: [asterisk-users] DIALSTATUS Values
 
 In article snt142-w54267269808afd17bccd5891...@phx.gbl,
 Kamlesh Kumar kamlesh_...@hotmail.com wrote:
  In addition to my reply:
  
  I used to fetch the value using print_r function but that also tells that 
  there is no value
  in data section.
  $dialstatus=$agi-get_variable(DIALSTATUS);
  print_r($dialstatus);
  
  SIP/10036-00b8AGI Rx  GET VARIABLE DIALSTATUS
  SIP/10036-00b8AGI Tx  200 result=1 (CANCEL)
  SIP/10036-00b8AGI Rx  Array
  SIP/10036-00b8AGI Tx  510 Invalid or unknown command
  [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() 
  returned error: Broken pipe
  SIP/10036-00b8AGI Rx  (
  SIP/10036-00b8AGI Tx  510 Invalid or unknown command
  [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() 
  returned error: Broken pipe
  SIP/10036-00b8AGI Rx  [code] = 200
  SIP/10036-00b8AGI Tx  510 Invalid or unknown command
  [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() 
  returned error: Broken pipe
  SIP/10036-00b8AGI Rx  [result] = 1
  SIP/10036-00b8AGI Tx  510 Invalid or unknown command
  [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() 
  returned error: Broken pipe
  SIP/10036-00b8AGI Rx  [data] =
 
 Well since the AGI return string does indeed contain the value, shown
 above as (CANCEL), that suggests there is definitely a bug in php-agi.
 It appears to be creating a ['data'] element, but not setting it.
 You will have to study the source code and work out how to fix it.
 I did a quick google for php agi get variable and found other reports
 of it not working properly, but I didn't see anyone offer a solution.
 It's only programming, so it shouldn't be hard to fix.
 
 Cheers
 Tony
 -- 
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 Work: t...@softins.co.uk - http://www.softins.co.uk
 Play: t...@mountifield.org - http://tony.mountifield.org
 
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Re: [asterisk-users] Using Asterisk as a softphone

[Sammy Govind]

Hi,
one reason for having that robotic voice could be improper codecs others
include low CPU processing power, memory not free etc. I once had the same
kind of issue with VAD(voice activity detection) turned ON from my service
providers equipment so my asterisk was performing poorly with VAD. Asterisk
version and its codec play more important role.

Regards,
Sammy

On Tue, Jan 3, 2012 at 6:34 PM, Christian Jaeger chr...@gmail.com wrote:

 Hello

 I'm using softphones as my only 'landline' phone service for almost 3
 years now (Diamondcard and now voip.ms), so far using SIP (and mostly
 Twinkle). Also, I'm using Linux (Debian) as my choice of desktop OS.
 Also, sometimes I'm in networks with badly behaving NAT routers (for
 some time I used openvpn to solve this unreliably, then I ended up
 using 3G instead of wifi while in Canada, but now I'm abroad and don't
 have 3G). I'm now sufficiently fed up with SIP to give IAX2 another
 try.

 I want a softphone solution that:

 * works on Linux (Debian)
 * works reliably (e.g. remain connected for incoming calls, work with
 shitty NAT routers)
 * preferably encrypts both signalling and voice (dunno if voip.ms
 supports it, I might use a proxy asterisk instance on an own server
 instead)
 * properly handles audio with the 8000 samples/second dictated by the
 POTS systems (ALSA combined with some hardware (like both of my
 laptops) doesn't do proper lowpass filtering for mic input, so I will
 have to either use OSS or PulseAudio or rely on Asterisk doing proper
 downsampling in software).

 Asterisk seems to fit the first three; I'll happily build a GUI on top
 if this turns out to be a stable solution.

 My problems right now:

 - when I issue console dial without a number, it plays a recording
 with a woman's voice, and I can understand what is being said, but it
 sounds very garbled, like modulated with some about 20 Hz signal (a
 bit like a robot voice). What could be the problem? (Not using
 pulseaudio; +- default configuration.) One hypothesis I have is that
 it uses a too small buffer somewhere.

 - I don't understand how the extensions stuff is working. voip.ms wiki
 told me to create sections named [voipms], but how do I switch to
 'default'?

 tie*CLI console dial 4443
 No such extension '4443' in context 'default'
 tie*CLI console dial 04443
 No such extension '04443' in context 'default'
 tie*CLI console dial 004443
 No such extension '004443' in context 'default'

 - I haven't found anyone in google who tried to do the same as me,
 except http://www.junghanns.net/en/asteriskassoftphone.html but that
 doesn't lead me far (and the patch linked is unavailabe). Has anyone
 here done what I envision, or seen some docs specifically matching my
 use case?

 Thanks
 Christian.

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Re: [asterisk-users] From address missing 'sip:', using it anyway

[Sammy Govind]

Hi,

The server or client application that is sending you sip packets is missing
the sip: string in from header. You should have it sorted out because if
that header goes to some external equipment the call may fail because of
this.

Regards,
Sammy

On Thu, Jan 5, 2012 at 12:44 AM, motty.cruz motty.c...@gmail.com wrote:

 Hello,
 I see the following error in the logs

 [Jan  4 11:37:35] NOTICE[21]: chan_sip.c:15388 check_user_full: From
 address
 missing 'sip:', using it anyway

 Does anybody know how to stop this error? It does not seem to be affecting
 performance on the Asterisk 1.8.4.1 running on Centos Linux 2.6. I have
 google it but empty!

 Thanks,
 Celso


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Re: [asterisk-users] DIALSTATUS Values

[Zohair Raza]

This works fine for me,

$dialstatus = $agi-get_variable(DIALSTATUS);
$cdr['dialstatus'] = $dialstatus['data'];

Try as it is, I believe it's because of concatenation.

Regards,
Zohair Raza




On Fri, Dec 2, 2011 at 4:27 PM, Tony Mountifield t...@softins.co.uk wrote:

 In article snt142-w54267269808afd17bccd5891...@phx.gbl,
 Kamlesh Kumar kamlesh_...@hotmail.com wrote:
  In addition to my reply:
 
  I used to fetch the value using print_r function but that also tells
 that there is no value
  in data section.
  $dialstatus=$agi-get_variable(DIALSTATUS);
  print_r($dialstatus);
 
  SIP/10036-00b8AGI Rx  GET VARIABLE DIALSTATUS
  SIP/10036-00b8AGI Tx  200 result=1 (CANCEL)
  SIP/10036-00b8AGI Rx  Array
  SIP/10036-00b8AGI Tx  510 Invalid or unknown command
  [Dec  3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write()
 returned error: Broken pipe
  SIP/10036-00b8AGI Rx  (
  SIP/10036-00b8AGI Tx  510 Invalid or unknown command
  [Dec  3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write()
 returned error: Broken pipe
  SIP/10036-00b8AGI Rx  [code] = 200
  SIP/10036-00b8AGI Tx  510 Invalid or unknown command
  [Dec  3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write()
 returned error: Broken pipe
  SIP/10036-00b8AGI Rx  [result] = 1
  SIP/10036-00b8AGI Tx  510 Invalid or unknown command
  [Dec  3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write()
 returned error: Broken pipe
  SIP/10036-00b8AGI Rx  [data] =

 Well since the AGI return string does indeed contain the value, shown
 above as (CANCEL), that suggests there is definitely a bug in php-agi.
 It appears to be creating a ['data'] element, but not setting it.
 You will have to study the source code and work out how to fix it.
 I did a quick google for php agi get variable and found other reports
 of it not working properly, but I didn't see anyone offer a solution.
 It's only programming, so it shouldn't be hard to fix.

 Cheers
 Tony
 --
 Tony Mountifield
 Work: t...@softins.co.uk - http://www.softins.co.uk
 Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] NAT/IPTABLES workarounds

[Sammy Govind]

Are you talking about having an SSH tunnel and route your SIP traffic
through it !!?

On Thu, Jan 5, 2012 at 4:20 AM, Kevin P. Fleming kpflem...@digium.comwrote:

 On 01/03/2012 10:03 AM, Patrick Lists wrote:

 On 03-01-12 16:24, Danny Nicholas wrote:

 Hello List,

 I work in an environment where I have to request IPTABLES changes rather
 than doing them myself. Is there a way to utilize my SSH (port 22) to
 get a functional (and with good sound) Asterisk installation with
 multiple channels up without requesting the 5060(SIP) 5061 (TLS) and
 UDP/RTP (usually 10001-2) IPTABLES allowances?


 Not with SIP as it needs a port for signaling (usually 5060) and RTP
 ports for sending the actual voice packets. So for SIP you will always
 need multiple ports. If you can use IAX then you could use port 22 as
 IAX only needs one port. The question is how are you going to SSH into
 the box if you use the SSH port for Asterisk?


 It is not practical (although not impossible) to run UDP over an SSH
 tunnel. Since VoIP media is generally transported over UDP, this will be a
 major obstacle.

 --
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 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org


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[asterisk-users] Where are the fax instructions?

[José Pablo Méndez Soto]

Hello,

Trying to set up res_fax_spandsp. Based on
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this in
my extensions.conf:

exten = 306,1,NoOp(Fax transmission)
same = n,Set(FAXOPT(gateway)=yes)
same = n,Dial(DAHDI/3)-FXS port to fax machine
same = n,Hangup()

Call flow Im trying to pull out is as follows:

Zoiper  --  Asterisk with TDM410 -- FXS -- Analog fax machine

I am totally lost about the use of this new gateway module in the dialplan.
I think it loads ok:

CLI fax show capabilities

Registered FAX Technology Modules:

Type: Spandsp
Description : Spandsp FAX Driver
Capabilities: SEND RECEIVE T.38 G.711

1 registered modules

Also I have the FFA manual, which I couldn't understand. I think FAXOPT is
common to both, but still not sure how to put them together. Where can I
find documentation about configuring the call flow described?

Or some insight will also be appreciated.

Here is my sip peer config:

[105](headquarters) ;zoiper phone
type=friend
secret=
mailbox=105@default
t38pt_udptl = yes

Dahdi:
;FXS Modules
group = 2
signalling = fxo_ks
context = interno
channel = 3-4
faxdetect = both

Finally, a verbose output:

  == Using SIP RTP CoS mark 5
-- Executing [606@intern:1] NoOp(SIP/105-0002, Fax
Transmission) in new stack
-- Executing [606@intern:2] Set(SIP/105-0002,
FAXOPT(gateway)=yes) in new stack
[Jan  5 00:59:57] WARNING[1831]: res_fax.c:2783 acf_faxopt_write: channel
'SIP/605-0002' set FAXOPT(gateway) to 'yes' is unhandled!
-- Executing [606@intern:3] Dial(SIP/605-0002, DAHDI/3) in new
stack
-- Called DAHDI/3
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 is ringing
-- DAHDI/3-1 answered SIP/605-0002
-- Hanging up on 'DAHDI/3-1'
-- Hungup 'DAHDI/3-1'
  == Spawn extension (intern, 606, 3) exited non-zero on 'SIP/105-0002'



Thanks in advance for any help

 *José Pablo Méndez
*
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Re: [asterisk-users] asterisk - AGI (perl) - sqlplus (oracle)

[LL]

I  guess this is a permissions issue.
Make sure your agi script has execute permissions (755)  and it belongs 
to asterisk:asterisk .


for that you need:
chmod 755 /var/lib/asterisk/agi-bin/agi-script-name.agi  chown 
asterisk:asterisk /var/lib/asterisk/agi-bin/agi-script-name.agi


Regards,
LL


On 1/4/2012 7:19 PM, Steve Edwards wrote:

Un-top-posting...

On Wed, 4 Jan 2012, Ahmed Munir wrote:

I'm trying to run an AGI in PERL which uses the module DBD-Oracle. 
Currently my AGI is working fine in my two servers but not in my 
other four servers. When I tried execute an AGI (as a user asterisk) 
in command line it works fine (even I also declare environmental 
variables in user profile and in my AGI), but when I tried to call my 
AGI (perl) in dial plan, it don't get executed.


It usually boils down to PATHs, environment variables, permissions, 
missing script interpreters, etc.


When you execute your AGI from the command line, are you passing a 
valid AGI environment to the AGI via STDIN? If not, you may be 
violating the AGI protocol, which, is probably not why your 'AGI' is 
not executing, but will bite you some time in the future.


When you say 'it don't get executed' do you mean that Asterisk cannot 
locate the AGI at all or do you mean it does not execute as you expect?


On Wed, 4 Jan 2012, Ahmed Munir wrote:

I installed the modules in asterisk user home directory with read and 
excitable permissions for asterisk but still my AGI not working.


Asterisk looks for AGI executables in the 'astagidir' (AKA ASTAGIDIR) 
directory which is usually /var/lib/asterisk/agi-bin/. This can be set 
in asterisk.conf. The path to this file can be specified on the 
command line used to start asterisk.


It would be unusual for ASTAGIDIR to be set the asterisk user's home 
directory. It would also be unusual for the asterisk user's home 
directory to be set to ASTAGIDIR.




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