regarding error in asterisk

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Hi all,
when i was trying to register a sipp client by using
register_client.xml file with .csv file in asterisk server i have
encountered an error that 1064 2150.240891 127.0.0.1 127.0.0.1 SIP Status: 481 Call
leg/transaction does not exist .... I dont kknow how does this error comes and i have searched whole the
internet i am unable to find a solution for this.....
Anyone please help mee.............. I am using asterisk 1.6.1......
version..... with sipp as client of 3.2 ......................
Awaiting for the as earliest as possible,

Asterisk Users 4.6 years ago 0 Answers

Streaming video on variable bandwidth connection?

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Hey guys, I'm sorry this isn't * related but if there *is* an 'answer' to this
question, I suspect someone on this list will know it. :) I'm trying to work out what technology to use; Situation:
Mobile Linux computer connected via 3G/GPRS to internet.
The computer is likely to encounter fluctuating connectivity where it
connectivity drops between low GPRS signal, full HDPSA signal and
completely offline. Objective:
I'm trying to find a technology to stream [live] video from a V4L2
device to 'the internet' over the able connection. The connection only

Asterisk Users 4.6 years ago 0 Answers

Newbie Question...

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Hello! Im new to Asterisk configuration and I have few questions regarding its
configuration. I have 4 PSTN lines connected to TDM410. I can make exact 60 minutes of free
calls from each of 4 pstn lines... Can I configure Asterisk to call thru
pstn line that has free minutes? For example Outgoing calls are going through PSTN 1 for 60 minutes. When I use all of
these free minutes - outgoing calls go thru PSTN 2. When I use all free
minutes from PSTN 2 outgoing calls go via PSTN3.

Asterisk Users 4.6 years ago 0 Answers

Calling Directory app from AGI

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Hi all, I've got an agi script that calls the directory function, which seems to
work to a point.  However, once the caller has selected an entry, I need my
agi script to find out which extension was selected.  I've RTFM'd and don't
see that the extension is returned.  Nor is a variable set, as far as I can
see. Is there a way to get this information from the directory application? TIA,

Asterisk Users 4.6 years ago 2 Answers

Issue with Asterisk not hanging up second leg when first leg hangs up

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Hi, Here is my confing: [out]
Exten => _X.,1,Noop()
Exten => _X.,2,Dial(SIP/${EXTEN}@peer,60,gcU(do_dtmf_cc-take-call,s,1))
Exten => _X.,3,Playback(tt-monkeys)
Exten => _X.,4,Playback(tt-monkeys)
Exten => _X.,5,Playback(tt-monkeys) Exten => h,1,Noop(ABCDEFGHIJKLMNOPQRSTUVWXYZ) [do_dtmf_cc-take-call]
Exten => s,1,AGI(agi://127.0.0.1:4579/update_call_status?status=60)
Exten => s,2,Noop()
Exten => s,3,Noop()
Exten => s,4,Noop()
Exten => s,5,Background(/etc/cb/wav/incoming_cb_call)
Exten => s,6,Noop()
Exten => s,7,Goto(s,5) Exten => _X,1,AGI(agi://127.0.0.1:4579/update_call_status?status=80) SIP Phone makes the call and calls an out side number. When out side number
picks up they hear the DTMF and then a message telling them to press any key
to take the call. If the…

Asterisk Users 4.6 years ago 0 Answers

Regarding error in asterisk or SIPp

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Hi every one,
I am using client as sipp and server as asterisk.... i
want to register an sipp client with asterisk..... I have configured
sip.conf and extensions.conf........ when i start asterisk and run sipp
register client.xml file i am getting result as register------------------10
401------------------------10
register-------------------10
200OK-------------------------------------------------10 as unexpected
message when i traced in wireshark and take a look at sipp -trace_err file i had
same error like : Aborting call on unexpected message while receiving 200 Ok
received 401---- unauthorized.. actually i have given .csv file also.......... i need…

Asterisk Users 4.6 years ago 0 Answers

invalid use of undefined type struct module

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All, I'm installing Asterisk with Dahdi on a server with a custom kernel compile.
I've got the kernel source in /lib/modules/2.6.34.6-xxxx-grs-ipv6-64/build
which points to /usr/src/linux-headers-2.6.34.6 and I think that's fine, but
am getting all these "struct module" errors. Can anyone advise? Thanks!
# make
make -C drivers/dahdi/firmware firmware-loaders
make[1]: entrant dans le répertoire «
/usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware »
make[1]: quittant le répertoire «
/usr/src/dahdi-linux-2.4.0/drivers/dahdi/firmware »
make -C /lib/modules/2.6.34.6-xxxx-grs-ipv6-64/build
SUBDIRS=/usr/src/dahdi-linux-2.4.0/drivers/dahdi
DAHDI_INCLUDE=/usr/src/dahdi-linux-2.4.0/include DAHDI_MODULES_EXTRA=" "
HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m
make[1]: entrant dans le répertoire « /usr/src/linux-headers-2.6.34.6 »
CC [M] /usr/src/dahdi-linux-2.4.0/drivers/dahdi/dahdi-base.o

Asterisk Users 4.6 years ago 2 Answers

Losing registration - ast 1.4.39 and innomedia 6328-2Re

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All, I'm having a problem with an Innomedia 6328-2Re (old Sunrocket Gizmo).
It keeps losing registration after a period of time ranging from a few
minutes to a few hours. It seems that right before it loses
registration, it fails to send a second register (after the 401
unauthorized). Here's a transcript from wireshark (at the end). The
last message is all that's received and asterisk now shows "UNKNOWN" ast
the status: 10676 17:13:05.255123 innomedia asterisk SIP Request:
REGISTER sip:asterisk
10677 17:13:05.255336 asterisk innomedia SIP Status: 100
Trying (0 bindings)

Asterisk Users 4.6 years ago 0 Answers