* You are viewing the archive for April, 2012

CONNECTEDLINE() updated during SIP events?

Hi,

I have read the excellent information here:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
and believe I have an understanding of what is offered. I have a
couple of questions:

– Is it possible to update COLP/COLR when a SIP redirect occurs, or
when a SIP divert is in place? If so, how?

– Is it possible to have the COLP/COLR information updated when a SIP
attended transfer is completed? If so how?

In both of the above cases, there is no obvious dialplan execution
when the calls are redirected, diverted or masqueraded, so we cannot
update the CONNECTEDLINE() information trivially. Or am I missing an
obvious trick?

Thanks,
Steve

Hangup Cause and SIP Response Code

I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?

Thanks

Bryant

Asterisk – Nortel transfer problem

Hi Carlos

How about to perform the call transfer operation (screened and unscreened) and trace the communication it will better on Nortel PBX side, in order to know which point is causing call transfer failure.
By the way these kind of features is complicated to handler on interconnected system, I just mean is no easy way for check intersystem resource state, as other than inband tones, mainly PBX system interconnect to others system by trunk resource and communication between system are handler by Trunk to Trunk PBX system control operation, PBX only use remote state (Busy, Congestion, etc.) for management own Trunk to Trunk resource operation.
Just in case of these kind of scenario it can be done through E1 (DR2 MFCR2 or DTMF) or ISDN PRI QSIG, lastone is more convenience because it cover more supplementary services than DR2 as; Completion of Calls to BusySubscriber (CCBS), Call Hold (HOLD)—by ISDN etc.
Best regards

Mc GRATH Ricardo
E-Mail mcgrathr@mail2web.com

Looking for IAX trunk/DID to replace Junction Networks

I received an email today from Junction Networks that they are
substantially increasing their monthly fee to the point that I’d be
cheaper getting a line from my local phone company. I’m now looking
for a replacement US carrier that supports IAX2.

I’m a home user and only need a single DID – preferably a local number
and only use a couple of hundred minutes of calls a month. All I need
is reasonably priced inbound and outbound IAX2 trunking, preferably
with the ability to set my CID.

If anyone has a low cost recommendation, I’d love to hear it.

Thanks,

Russell.

Asterisk – Nortel Transfer Problem

I have an Asterisk server connected to a Nortel Pbx via an E1.  Everything works fine, I get calls in and out with callerid. The problem that has been reported to me is the following scenario:

A call comes in from the PSTN and is answered by Asterisk. The person dials the operator (1000) which is on the Nortel side so connection is made through the E1. The operator answers  and then transfers the call back to a SIP extension on the Asterisk (1303). The result is no audio and a dropped call.

My main theory at the moment is that when the receptionist hangs up after the transfer the E1 drops on the Nortel side. Anyone here with  this type of integration seen this problem?

Strange problem on ougoing call

Hi

i have a strange problems on my asterisk server:

I have two asterisk server.

On the first, i use realtime with a MySQL Database,
i have two user:
USER01
USER02
exactly the same configuration only username and password has different.

On my second server (phone is connected on this server):

I have in sip.conf:

register => USER01:1234@172.16.0.11/USER01
register => USER02:5678@172.16.0.11/USER02

[USER01]
type=friend
username=USER01
secret=1234
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite

[USER02]
type=friend
username=USER02
secret=5678
host=172.16.0.11
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=no
dtmfmode=rfc2833
disallow=all
allow=alaw
context=I-User01
musiconhold=default
insecure=port,invite

i see the registration:

ipbx*CLI> sip show registry
Host dnsmgr Username Refresh State
Reg.Time
172.16.0.11:5060 N USER01 105 Registered
Tue, 24 Apr 2012 15:58:58
172.16.0.11:5060 N USER02 105 Registered
Tue, 24 Apr 2012 15:58:59

i have one phone connected to the context “I-User01″ and another
connected to “I-User02″

When i call with the phone connected to I-User01, no problems, that’s
work but when i call
with the second phone (use I-User02) i have a error:

On the first server:
[Apr 24 16:02:12] WARNING[1456]: chan_sip.c:10992 check_auth: username
mismatch, have , digest has
[Apr 24 16:02:12] NOTICE[1456]: chan_sip.c:18096
handle_request_invite: Failed to authenticate device “Olivier”
;tag=as0cd775ab

The exten:

On I-User01: exten => _0XX.,1,Dial(SIP/USER01/${EXTEN:1},90,r)
On I-User02: exten => _0XX.,1,Dial(SIP/USER02/${EXTEN:1},90,r)

i i change on the I-User02:
Dial(SIP/USER02/${EXTEN:1},90,r)
in
Dial(SIP/USER01/${EXTEN:1},90,r)
all call work’s.

anyone have a idea ? i think’s that i have a error but don’t see where

best regards
Olivier