I’m attempting to pull SIP users from LDAP, following the instructions from here:

http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ExternalServices_id291590.html

However, when I attempt to register a user from LDAP, I see on the console:

chan_sip.c:24431 handle_request_register: Registration from ‘‘ failed for ’172.20.24.131:3072′ – No matching peer found

Poking around with tcpdump I see that Asterisk sends a query to the LDAP server, and gets a response for that user, so the connection is working at least. So, Asterisk must not be recognizing the record as valid. However, I have no idea why. Is there some more verbose logging that can be enabled? How can I do test lookups with the realtime engine? I can’t find any place to start troubleshooting this issue.

I’m full of other questions, like:

what’s the difference between “sipusers” and “sippeers” in extconfig.conf?

what happens if I don’t provide all of the attributes (amaflags, callgroup, callerid…there are dozens listen in res_ldap.conf) in my LDAP directory? I don’t really want to duplicate all those settings for each user in my organization: that’s not very managable.

Any guidance from someone who’s been down this road would be appreciated.

Hi,

I’ve read here and there how Asterisk could send SMS but I didn’t find
much about how to receive SMS and forward them to an email box.

1. First of all, I don’t think my telco would let me receive any SMS
my landline.

2. Maybe I could find providers selling this service for a monthly fee;

3. I could build and operate my own infrastructure.

Given this asterisk-users mailing-list purpose, and for curiosity’s
sake, how could I build my own SMS reception service with Asterisk
(1.6.1 or later) ?
Which channel (chan_mobile, chan_datacard, …) and hardware would be
appropriate ?
Suggestions ?

Regards

Hi All,

I just installed Asterisk 10.1.1 and on sip reload I see the following two errors.

WARNING[3665]: chan_sip.c:29242 reload_config: No valid transports available, falling back to ‘udp’.
== Using SIP CoS mark 4
WARNING[3665]: chan_sip.c:27839 build_peer: ‘tcp’ is not a valid transport type when tcpenabled=no. If no other is specified, the defaults from general will be used.

I don’t understand this as I have in sip.conf
udpbindaddr=0.0.0.0

I’ve also tried
udpbindaddr=192.168.13.7
and
udpbindaddr=192.168.13.7:5060

with the same results.

Is there another setting that I am missing. I have not made any changes to the default file settings, well none that I remember.

Regards
David Klaverstyn
Systems Engineer
Information Services, Asia-Pacific
Intergraph Corporation
270 Gladstone Road
Dutton Park, QLD 4102 AU
P 61.7.3510.8951 F 61.7.3510.8980
david.klaverstyn@intergraph.com, www.intergraph.com.au

Hi,

I have a Junghanns QuadBri installed and seems to be detected and working
correctly using the wcb4xxp driver
All ports are configured as TE

The problem is when I connect an ISDN line – the LED turns green and Dahdi
status shows Alarm OK
But when I try to make a call I get “All circuits are busy” cause 27

I have searched and web and tried numerous configs but it doesn’t want to
play.
I can accept incoming calls but there is no Audio which would indicate an
issue with the driver wcb4xxp?

See config files below and dahdi commands – please ignore that RED alarms
there was no ISDN lines plugged in.

Thanks, Joe

Here is CLI from a call

[2012-02-03 08:54:25] VERBOSE[13586] app_dial.c: — Called DAHDI/g0/1901
[2012-02-03 08:54:25] VERBOSE[13586] app_dial.c: — DAHDI/3-1 is proceeding
passing it to SIP/221-00001c80
[2012-02-03 08:54:27] VERBOSE[24518] chan_dahdi.c: — Channel 0/3, span 1
got hangup request, cause 27
[2012-02-03 08:54:27] VERBOSE[13586] app_dial.c: — DAHDI/3-1 is
circuit-busy
[2012-02-03 08:54:27] VERBOSE[13586] chan_dahdi.c: — Hungup ‘DAHDI/3-1′
[2012-02-03 08:54:27] VERBOSE[13586] app_dial.c: == Everyone is
busy/congested at this time (1:0/1/0)

On 01/28/2012 10:22 AM, Din Assegaf wrote:
> Hi All,
>
> I’m trying to upgrade asterisk server to 1.8.x from my asterisk 1.6,
>
> But when making A Call from SIP Client, I got cli Warning … and no
> call has been made.
>
> My Sip Client is using lib java peers client http://peers.sourceforge.net/
> with standard codec PCMU/PCMA
>
> [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:8942 process_sdp:
> Unsupported SDP media type in offer: audio 0 RTP/AVP 0 8 101
> [Jan 28 23:03:32] WARNING[1654]: chan_sip.c:9029 process_sdp: Failing
> due to no acceptable offer found
>
> the strange thing is when using asterisk 1.6, is normal,
> when using asterisk 1.8.x and using another client like Ekiga is normal too,

The error message is misleading; you are having this problem because the
‘m’ line in the SDP with the ‘audio’ offer has a port number of 0
(zero)., which means it is not an active media stream offer. It does not
make any sense for the SDP in an INVITE for a new call to have an m-line
with a port number of zero.

I’ll improve the error message so that this sort of situation won’t be
as confusing in the future.