Intermitent voice issues

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Hi all and thanks for reading. I am experiencing a frustrating issue with asterisk where on some
calls the volume suddenly drops to inaudible o completely fades away
for a time. If you hold on long enough (20 to 30 seconds) the sound
will come back. My asterisk server is on a public IP, and basically acts as a VoIP
bridge receiving calls from my customers (all of whom use Grandstream
GXW400X gateways on public IP's, no NAT) and sending them to different
SIP providers. I am proxying the RTP stream through the server

Asterisk Users 4.6 years ago 0 Answers

Caller ID

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_____ From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Cary Fitch
Sent: Tuesday, March 01, 2011 11:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Caller ID We do not get caller ID (name) on our telco lines. However we have a few single line extensions with consumer type handsets
that ring at odd hours with "Asterisk" before the phone is picked up, and
"Out of Area" after it is picked up. I have read that "Asterisk" is what is reported by Asterisk for 0 length
caller ID number. But since…

Asterisk Users 4.6 years ago 0 Answers

records inbound and outbound calls

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Hello List i have asterisk installed in our call centre i have configured the snom
phone 320 and 370 with in sip.conf and dialplan.com and extenssion.com i have just one question how can i do in order to record all the calls
automatically in our server Thanks and regards

Asterisk Users 4.6 years ago 5 Answers

IRQ 0 on BRI card (B200E)

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Hello list, I'm having a problem with a BRI card, a B200E (PCI-e) card on an Intel
server with a riser that I'm guessing is what's making the problem. The output of lspci -vvv is: 07:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network
Controller [HFC-4S] (rev 01)
Subsystem: Cologne Chip Designs GmbH Device e884
Control: I/O+ Mem+ BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr-
Stepping- SERR- FastB2B- DisINTx-
Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort-
SERR- Interrupt: pin A routed to IRQ 0
Region 0: I/O ports…

Asterisk Users 4.6 years ago 0 Answers

Simple way to bridge two channels?

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Hello I'd like to know what my options are to bridge two channels after
calling each through Dial(). I know about MeetMe, Conference, and Konference. Are there other
options available just to bridge two calls? www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe
www.voip-info.org/wiki/view/Asterisk+cmd+Conference
www.voip-info.org/wiki/view/Asterisk+cmd+Konference I'd like the simplest possible solution since Asterisk is running on a
non-x86 appliance, so code must be cross-compiled. Thank you.

Uncategorized 4.6 years ago 0 Answers

two questions regarding incoming call

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Hello,
I want to make an agi script to match incoming DIDs with usernames. I tried to do such entry in incoming trunk. [DID_diddw]
include = from-didww [from-didww]
exten = 3130XXXXXXX,1,AGI("did.php")
exten = 3130XXXXXXX,n,DIAL(SIP/${yup_no},20)
but when i run the rule it says
chan_sip.c:20152 handle_request_invite: Call from '81.85.224.41' to extension
'31301111111' rejected because extension not found in context 'from-didww'
Cant I use such agi scripts on incoming calls? PS:
exten = 3130XXXXXXX,n,DIAL(SIP/1111) works alone.
My second question.
I got two incoming trunk sip channels on my server. One…

Asterisk Users 4.6 years ago 1 Answer