* You are viewing Posts Tagged ‘SoftSwitch’

Hosted Softswitch Integration

Hello Everyone,

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.


remote UPDATE command

Hi guys,
I am working on a setup where I have an Asterisk ( 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..


SS7 + T1

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
so forth?)?

Asterisk as SoftSwitch – Hardware

Hi list,

Could anyone tell me what is the “recommended” hardware to a system for
following configuration:

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,

Failover trunks

Hi All,
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.
Please help.
Regards————-Abid Saleem

How to reload queue on the fly?


On the production server I’ve modify the /etc/asterisk/queues.conf file. Now
in CLI I wan’t to reload queue configuration gracefully. I did:

virtual-pbx*CLI> queue reload members office

But `queue show office` tells me that nothing has changed. I tried to reload
all — `queue reload all':

virtual-pbx*CLI> queue reload all
[Dec 28 15:10:01] NOTICE[25405]: app_queue.c:5668 reload_queue_rules:
queuerules.conf has not changed since it was last loaded. Not taking any

And still my configuration is not applied.

Current queue for `office':

virtual-pbx*CLI> queue show 1telecom_office
1telecom_office has 1 calls (max unlimited) in ‘linear’ strategy (0s
holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
SIP/121 (Ringing) has taken no calls yet
SIP/120 (Not in use) has taken no calls yet
SIP/123 (Not in use) has taken no calls yet
1. SIP/Softswitch-000161ae (wait: 0:01, prio: 0)

While modified configuration is:

strategy = linear
timeout = 10
member => SIP/100
member => SIP/101
member => SIP/121
member => SIP/123
member => SIP/120
monitor-format = wav
monitor-type = MixMonitor
joinempty = yes

What’s may be wrong?