We are trying to integrate a hosted soft-switch to an Asterisks server and the error received on the Softswitch end is decline 603
The change that we made is to add the Softswitch IP in the SIP
configuration file, see below
type=user nat=yes insecure=very dtmfmode=rfc2833
context=from-trunk canreinvite=no disallow=all allow=ulaw allow=gsm allow=g729
On the attempt to integrate to the asterisks server nothing is seen in the asterisk log There is something we might not be doing well. can somebody please help.
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I am working on a setup where I have an Asterisk (18.104.22.168) with a SIP trunk
to a provider’s softswitch(IMS). Trunking works all ok, calling out and
calling in. Except for the remote softswitch(IMS) will send UPDATE
command(according to RFC 3311) and Asterisk does not accept and thus the
call gets dropped.
So, is there a work around for this matter? Or any of you had similar
problem, I really appreciate any directions.
The interesting part is that, they have an appliance based Asterisk that
works perfectly fine, so I assume that it actually works ok. Just need to
find out hows that possible..
I spoke with the Asterisk Pre-sales team and they said that SS7
support isn’t technically supported, but it is there (e.g. talk to the
OS community about this)…. so here’s my question:
I’m trying to interface an Asterisk Softswitch to a Nortel DMS100.
If I get a dual-span card, can I run SS7 signaling over one span, and
a T1 over the other span and have Asterisk link the two (e.g.
caller-ID for the call on the 1st channel comes across the SS7… and
Could anyone tell me what is the “recommended” hardware to a system for
SBC –> Asterisk (SS) –> Carrier GW
Asterisk should work as a Class 4 SoftSwitch, with following functionalists:
-> Do the IP Authentication
-> All communications on RTP/G729 (no transcoding required)
-> Load of 1200 concurrent call sessions
-> No call routing required
Thanks in advance,
Could you please help me with my following Scenario. I have a softswitch where my carriers send calls from International to my country for local termination. I route these calls to my Asterisk 1.8 which has a number of registered trunks from our SIP Provider. Please guide me how should I configure extensions.conf for calls to be sent to the next available trunk.