Displaying 20 results from an estimated 90000 matches similar to: "SIP/REGISTER problems!"
2003 Nov 07
2
No ringing tone
I have the following setup:
AnalogPhone1--TDM400P-ASTERISK---via SIP---Softswitch--------POTS Phone2
When I call from AnalogPhone1 to Phone2 I hear a ringing tone and all is well.
When making a call from Phone2, I get a dial tone but after dialing the number
I hear nothing (no ringing tone). On Asterisk console it says that a call is
coming in and that it is ringing Zap/2. I can also hear the
2003 Nov 06
6
Error in Incoming SIP call
When I get a SIP call, I get the following error:
*CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is
'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp'
WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No
application '' for extension (incoming, 5147771111, 1)
== Spawn extension (incoming,
2003 Nov 07
1
No communication channel
I have following setup:
AnalogPhone_1--TDM400P--Asterisk---SIP---[Softswitch]----POTS-AnalogPhone_2
I can call from AnalogPhone_1 to AnalogPhone_2 and all is fine.
When I call to AnalogPhone_1 from AnalogPhone_2, AnalogPhone_1 rings BUT
I hear no ringing tone AND when someone picks up AnalogPhone_1, there is no
"sound" and parties on both end cannot hear each other. Seems that no
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?
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
2008 Apr 02
2
Howto connect to Cirpack softswitch with Asterisk ?
Hi,
has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto
or more info about needed Asterisk SW and setup ?
Thanks in advance,
regards,
Rob.
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
2003 Oct 17
4
Using channel banks
Hello Everyone,
What kind of hardware setup would I need to do if I want a T1 connection to PSTN
and have 48 users in office with analog phones. Will something work if I have a
T410P card in asterisk and have one T1 going to PSTN and other two to a channel
bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks.
Deepak
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
2006 Oct 17
0
lots of registrations, sip problem
Hello,
I've got a problem with connection to my SIP provider. In general,
everything works, but I get lots of these messages:
Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's
odd... Got a response on a call we dont know about. Cseq 42710 Cmd
SIP/2.0
Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request:
That's odd... Got a response on a call
2007 Oct 19
1
Using register => to let Asterisk register to another softswitch via SIP
Hi All;
Alot of softswitches or PBX's does not accept to
manipulate any SIP call without being registered
firstly. So that means, I need asterisk to register
firstly then I can route my calls to that SIP trunk.
In IAX2, we use the register => , so what shall we do
in Asterisk? And how its format will be (if we will
use register)? Or what is the solution?
Regards
Bilal
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
2005 Sep 20
0
Handling SIP 404 event
Hello all,
I am curious, does anybody know of a way to handle the SIP 404 event?
(ie: is this stored in a variable somewhere, so one can handle it in
the dial plan).
For example, dialing an invalid number on another softswitch on the network:
-- Executing Dial("SIP/sip7110-8118", "SIP/7234@softswitch|60|r")
in new stack
-- Called 7234@softswitch
-- Got SIP
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
2005 Mar 04
0
TE405P and quality problem
Hi,
I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch
(www.cirpack.com).
<IP Network>--<*>-[TE405P]-<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
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 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
2008 Feb 11
1
SIP Bad request protocol Packet on Asterisk 1.4.18
Hi all!!
I have a really weird problem. I upgraded 2 Asterisk 1.2 boxes to Asterisk
1.4.18. Both are home PBX's and both boxes register to a SIP DID at
exactly same provider. One box runs without errors on the console, the
other box keeps repeating :
[Feb 11 23:40:29] WARNING[11292]: chan_sip.c:6705
determine_firstline_parts: Bad request protocol Packet
When i set debug on, it seems to
2007 Jan 03
2
Error on answer a SIP 401 message
Hi,
I'm a voip service provider and i'm setting up a asterisk box to
register around 100 lines from my central softswitch. This asterisk
box will be placed inside a customer and has a digium card to be
interconected with customer's pabx.
My problem is that when asterisk send register message, my softswitch
return with sip 401 and asterisk should send a register message with
2004 Dec 17
0
AS5xx0: SS7 and SIP?
We currently use Asterisk to provide a SIP-to-PSTN service. The actual
conversion takes place somewhere in a softswitch owned by our
SIP-to-PSTN provider, where we have an SS7 link. We would like to do
that conversion ourselves.
Is it possible to replace a softswitch with a Cisco AS5xx0 only (ie.
AS5300, 5350, 5400), or is a *real* softswitch (ie. Cisco PGW2200)
needed? Does anyone have any