how to use qualify times to route calls

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I'm using 1.8.3, and have 2 sip providers. Both are set with
qualify=yes. Each of them sometimes have qualify times 10+ times the
other. For instance, one will be at 10-15ms, the other at 200ms. Is there a way I can route an outgoing call to the provider with the
lower qualify time? sean

Asterisk Users 4.6 years ago 1 Answer

asterisk behind nat

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On 3/2/2011 9:46 AM, Leif Neland wrote:
> Some of the phones are being disconnected with Asterisk saying "no reply
> to critical packet" What kind of phones are they? I might have nothing to do with your
network configuration; try adding to sip.conf [general]: session-timers=refuse

Asterisk Users 4.6 years ago 1 Answer

Doubt about cdr on asterisk

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I have the following situation.... I'm using Action Originate to originate a call for a costumer. Originate goes to a context that call the dial application. Before the application (Dial using the G option) to be invoked i'm setting the variable cellphone like this: [firstcontext]
exten => s,1,Set(CDR(cellphone)=${CELLPHONE})
exten => s,n,Dial(IAX2/user:pass@otherasterisk/${CELLPHONE},30,G(secondcontext^s^1)) [secondcontext]
exten => s,1,Hangup()
exten => s,n,Playback(something)
exten => s,n,NoOp(CDR(cellphone)
exten => s,n,Hangup() When the costumer answer the call, caller party goes to secondcontex on extension 1 and the called party goes to secondcontex on extension 2. On firstcontext (before the Dial) i…

Asterisk Users 4.6 years ago 1 Answer

Registering Cisco 7942G IP phone with Asterisk!.

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Hi,
 
We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)).
 
Recently we have bought a cisco 7942G IP phone.
It currently has SIP 42.9-0-2SR1S firmware loaded on it.
We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)  IP address (with current firmware on it) to register it with Asterisk.
 
Do we need to upgrade the SIP firmware to any latest versions?
If yes, to which version we should be updating…

Asterisk Users 4.6 years ago 0 Answers

Asterisk 1.8 SIP realtime and NAT

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Hi After recently upgrading to 1.8.3 I have noticed that the nat setting
for my peer in my sip table is not making it into the realtime cache. For example
* Name : 501
Realtime peer: Yes, cached
Secret :
MD5Secret :
Remote Secret:
Context : pack-local
Subscr.Cont. :
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
MOH Suggest :
Mailbox : 501@local
VM Extension…

Asterisk Users 4.6 years ago 1 Answer

GSM-Card for Asterisk / recommendation needed

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Hi, I am trying to setup a GSM-Card for Asterisk. I currently use a "vgsm I"
from voismart (http://www.voismart.it/) but the driver is very bad
(compile-problems and no echo cancellation). Is there anybody out there who can recommend me another piece of
hardware (pci card)? I need 1 or better 4 gsm-ports. Should be stable
and have an echo cancelltaion feature. And of course it should be cheap ;-) Best regards
-Thorsten-

Asterisk Users 4.6 years ago 1 Answer