Hi There,
I’m fairly new to asterisk and I’m trying to play a video file during a
video call without success for a couple of days now.
I’ve posted a question at stack-overflow describing my problem -
http://stackoverflow.com/questions/8675713/asterisk-video-playback-mp4-3gp
*
In short, what are the exact specs that a 3gp or mp4 file needs to have
in order to play under asterisk ?*
Any tips, links or suggestions will be very welcome.

*Thank you all!
LL
*

Greetings,

Is there a way to block a specific inbound number? I’ve found code online for blocking all nocallerid and all 800, etc. but nothing for a specific number. My company is wanting me to block a specific number. Is this possible in Asterisk 1.4 and 1.6 or do I need to go through my Service Provider?

Thanks,

Kevin Oravits
Phone Sys Admin

Dears,

1-I have a GSM gateway (GOIP) with 8 ports, I used to let every port
register to VoIPSwitch in order to know how many minutes does this GSM
card, ASR ,ACD on each card.

It’s too simple on VoIPSwitch to add the registrar client to dial plan ,but
in asterisk only I can find trunks

How can I do that with asterisk .

2-Do any one know from where I can download a2billing prompts in Arabic for
free.

Regards

Khaled Chehab

NGN Eng.

Description: xplorium

Operations Office – Lebanon

Office : +961 1 868686 ext 115

Mobile: +961 3 045212

E-mail: kchehab@xplorium.com

MSN ID :KhalidChehab@hotmail.com

Web Site: http://www.xplorium.com

winmail.dat

Trying to set up a simple sip – sip connection over tcp. Home : 10.0.0 -
Office: 1.8.8.0

Home sip.conf:

register =>
tcp://office-going-to-home:password@/home-coming-from-office

[home-coming-from-office] ; receives calls
type=friend
transport=tcp
dtmfmode=rfc2833
disallow=all
allow=ulaw
secret=password
context=incoming

Office:
[office-going-to-home] ; places calls
type=peer ;; we only call out
transport=tcp
dtmfmode=rfc2833
disallow=all
allow=ulaw
fromuser=home-coming-from-office
remotesecret=password

sip show peer office-going-to-home

* Name : office-going-to-home
Secret :
MD5Secret :
Remote Secret:
……..
Status : OK (28 ms)
Useragent : Asterisk PBX 10.0.0
Reg. Contact : sip:home-coming-from-office@ ipaddr>:5060;transport=TCP

But if I place a call from the office to home, on home cli:

[Dec 29 16:32:23] WARNING[4387]: chan_sip.c:14446 check_auth: username
mismatch, have , digest has
[Dec 29 16:32:23] NOTICE[4387]: chan_sip.c:22769 handle_request_invite:
Failed to authenticate device “Office”
>;tag=as150d4960

Hello,

I’m trying to interface Asterisk with a third party voicemail system. This
voicemail system registers itself as extension 199. This voicemail system
gets the DID number (mailbox) from the SIP To: header field.

My problem is creating the SIP INVITE with a To: field that’s different
that the Request-Line URI

I need to create an INVITE that looks like this in Asterisk:

20:23:12 UDP Packet Received from 127.0.0.1:5060
< <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<

INVITE sip:199@127.0.0.1:5070;target=101%40voicemailserver;cause=486 SIP/2.0

Via: SIP/2.0/UDP 10.11.11.205:5060;rport;branch=z9hG4bK253292

To:

From: “101″;tag=3277

Call-ID: 1325031862-3292-VOICEMAILSER@10.11.11.205

CSeq: 14 INVITE

Can this be accomplished?

Mike