Displaying 20 results from an estimated 2000 matches similar to: "Howto connect to Cirpack softswitch with Asterisk ?"
2007 Dec 29
2
Cirpack KeepAlive packets causing SIP errors
Hi list,
After a recent upgrade to Asterisk v1.4.14, my message log is now
filling up with
the following error messages:
<------------->
[Dec 29 17:24:52] WARNING[10655]: chan_sip.c:6645
determine_firstline_parts: Bad request protocol Packet
--- (1 headers 0 lines) ---
bitis*CLI>
<--- SIP read from 82.101.62.99:5060 --->
Cirpack KeepAlive Packet
<------------->
Seeing
2008 Mar 20
1
423 "Interval Too Brief" and expiry settings in sip.conf
Hi,
I'm getting this error when registering with SIP server using Asterisk
1.4.10 and Freepbx...
I'm getting this error no matter what I try to setup in sip.conf :
- I'm getting confused whether options are maxexpirey=36000 or
maxexpiry=36000 ?
- Can I solve this with some settings in sip.conf or is this problem harder
?
- I've read something about Asterisk's bug on this
2004 Oct 05
2
SIP multipart mime messages
I was messing about integration of a Cirpack softswitch with Asterisk
and banged my head against a problem previously noted on the list.
http://lists.digium.com/pipermail/asterisk-users/2003-November/026436.ht
ml
What is the status of this problem? Has it been fixed? I scrambled
through chan_sip.c, but couldn't find ay reference to "multipart".
Regards,
Jesper Dalberg
2005 Feb 23
0
Digium TE405P and Cirpack Switch
Hi,
I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch
(www.cirpack.com).
<IP Network>--<*>--<Cirpack>--<Public PSTN Network>
ISDN interco is EuroIsdn, ports are configured ccs, hdb3, crc4. Cirpack
is Network, * is Terminal/User.
As I encountered some pb with Sip to Zap transcoding (* to Cirpack way
poor quality, the other way fine), I tryed to
2005 Feb 28
0
Pb DTMF with Asterisk vs Cirpack Transit Node
Hi all,
I've a problem in DTMF dialog between * and a Cicpack Transit Node Class 4-5.
The call initiat by a mgcp phone pass by the cirpack and arrive in SIP on *,
everything is ok (negociation and phone call) but when we try to use the
voicemail, Asterisk don't understand DTMF.
Here are some logs (SIP debug on) on a DTMF '2' receive :
2005 Feb 28
0
Pb DTMF with Asterisk vs Cirpack Transit, Node
Salut Guy,
I have the same problem with a Cirpack (B3G carrier)
What I see is that you use sip info to detect DTMF.
The problem is that there is no normalisation on the content of the sip
info frame for dtmf detection.
First, asterisk try to detect the header "application/dtmf-relay"
and you have the header "application/dtmf"
see line 6069 of /channels/chan_sip.c function
2005 Mar 01
0
RE: Pb DTMF with Asterisk vs Cirpack Transit, , Node
Salut Guy,
I have the same problem with a Cirpack (B3G carrier)
What I see is that you use sip info to detect DTMF.
The problem is that there is no normalisation on the content of the sip
info frame for dtmf detection.
First, asterisk try to detect the header "application/dtmf-relay"
and you have the header "application/dtmf"
see line 6069 of /channels/chan_sip.c function
2009 Jan 14
2
Any free video (or audio) softphone VOIP client under Linux with touchscreen friendly interface ?
Hi,
I'm curious if anyone knows of any possibility to use video VOIP client
(like Ekiga or Linphone or...) under Linux that could be operated by
touchscreen friendly GUI (bigger buttons, large keypad, etc...) ?
I like Ekiga, but GUI is small and cannot be operated via touchscreen... But
maybe there are some skins for existing clients that are more touchscreen
friendly ?
Thanks in
2006 Mar 06
1
most common VOIP echo simulaton for research purposes ?
Hi,
I'm speech recognition researcher and would like to do some research on
recognition robustness in echo distortion of speech signal. Since VOIP is
becoming wide spread, I'd like to simulate (one or more) common echo
distortions that mostly appear in voip communications ? Any example, FIR or
IIR filter or acoustical system response ?
Any other distortion worth researching ?
Thanks
2008 Mar 18
1
Using dedicated eth2 card for SIP trunk line to ISP provider - how to setup ?
Hi,
I'm about to test VOIP connection (from my ISP provider) directly through
dedicated network card instead of going through ADSL gateway with analog
phone port - SPA 3000 - Asterisk.
I need to have eth2 set on dhcp (to retrieve IP automatically) and then work
with it under Asterisk as dedicated VOIP trunk.
Anyone with more insight how to setup such situation ? Any more info
anywhere
2005 Feb 18
5
Which PRI card for EuroISDN ?
Hi,
I wonder which PRI interface card is most stable and supported for EuroISDN
and Asterisk ? Are they stable enough ? Any tips ?
Thanks in advance,
regards,
Rob.
2007 Jul 24
2
SIP IP Trunk, between Asterisk and Softswitch
Dear List;
I am trying to create a link between Asterisk and My
softswitch, the link to be SIP Trunk.
I did the below configuration and I do not know if any
one can help me and advise me to have better
configuration to be sure that link is fine. But I do
not know how to determine the SIP username to be sent
for my softswitch as sometimes the softswitch need to
check it.
Also, does asterisk
2007 Sep 19
2
what is softswitch
Dear all
what is softswitch what is difference between asterisk and softswitch ??
regards
satish patel
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2005 Jan 20
4
softswitch dilemma
Hello everybody,
Im new to the list and also new to asterisk, Im wondering if I could set up asterisk as a softswitch, I guess for what I've been reading that It could be possible but almost all the info and documentation Ive found so far is about asterisk as a PBX, etc.
Im willing to set a small voip wholesale traffic bussiness and Im not quite sure asterisk is the right chocie for that.
2007 Dec 02
2
Softswitch digim
Hello averybody,
I'm looking the softswitch in digium website, anyone test the softswitch?
Best Regards
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2007 Oct 23
2
register => to let Asterisk register to another softswitch via SIP
Dear Alex;
Thanks alot for your nice help.
This is if I need to let Asterisk register with
another softswitch (so I used register =>), what if I
need asterisk to send call for the softswitch without
register to it (directly)? If I removed the register
=> then how it will distiguish the IP address in the
"host" at the [sip_trunk] is the IP address of the
softswitch that need to
2007 Jun 28
2
Linking Asterisk with another SIP PBX (or SIP Softswitch)
Hi List;
If I need to do a trunk between Asterisk and another
SIP softswitch (so Asterisk will send a SIP calls to
that softswitch), then I have to configure this on the
sip.conf file or where exactly? And is it the same
when I configure iax trunk?
Should I determine the context in this case for this
SIP trunk?
Regards
Bilal
2003 Oct 24
8
SS7 signaling/Softswitch
I'm confused a bit about the following and was hoping to get some answers on
this group - What is exactly implied when we say asterisk can connect to a PSTN.
Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume
asterisk does not need to do any SS7 signaling and all it does (playing the role
of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct
statement?
2003 Aug 06
2
Semi-newbie question "Softswitch" and Asterisk - Is there a difference?
I've been working in the VoIP industry for just a bit over a year now...
Mostly taking care of the underlying systems. I've now reached the
point where I'm being drawn more and more into the call processing side
of things. My background is in computer and "classic" telephony systems
(DMS250/MTX, DSC 400, T1, channel banks. telabs analog echo supressor
modules and
2003 Nov 07
2
Softswitch
Pardon my ignorance, but I was hoping someone could clear up something for
me.
- For a few POTS lines, digium has a single port card for that, or a T1 card
to a channel bank.
- For 10 or more lines, digium has a T1 or E1 card for that too based on PRI
channels
- For 100's to 1000's of lines, I suspect a soft-switch is in order???
A traditional phone company will sell:
- POTS lines for