DAHDI spans up without physical connections

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We are in the process of rebuilding our servers and upgrading asterisk and
zaptel/dahdi. The server I'm currently working on was:
OS: Fedora fc9
libpri 1.4.10.2
Zaptel 1.4.12.1
Asterisk: 1.4.27 After rebuild:
OS: CentOS 5.7 (Final) installed via Spacewalk
libpri: 1.4.12
DAHDI: dahdi-linux-complete-2.5.0.2+2.5.0.2.
Asterisk 1.8.6 Digium card TE420:
pci:0000:09:08.0 wct4xxp+ d161:1420 Wildcard TE420 (5th Gen) dahdi spans are configured as:
span=1,0,0,esf,b8zs
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs fxoks=1-24
fxoks=25-48
fxoks=49-72 No errors during build process, but the dahdi spans are showing up even
though there are no cables…

Asterisk Users 3.9 years ago 0 Answers

Check which client access Asterisk using AMI

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Hi, In manager.conf file I created a user profile by which clients can access
Asterisk server as listed below;
[cbusapp]
secret = cbus123
deny=0.0.0.0/0.0.0.0
permit=192.168.1.0/255.255.255.0
read = system,call,log,verbose,command,agent,user,originate
write = system,call,log,verbose,command,agent,user,originate Using above configuration clients are successfully access the asterisk and
forward its parameters to asterisk. The thing I would like to know how can I
keep track from which client does asterisk receives request from? Like
client A, B and C need to know from which clients the request was made to
asterisk.

Asterisk Users 3.9 years ago 1 Answer

Tips & best practices for asterisk troubleshooting & parsing logs

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Capture pcap with tshark or tcpdump for the future analysis with wireshark.
Ngrep is also handy tool for captaring, say, INVITE. You can use grep like
this: tail -f /var/log/asterisk/full | egrep --color -w
'chan_sip.*SIP/911|pbx.*SIP/911'
Interesting technique from Astresk Cookbook, "Debugging dialplan with
Verbose()
http://ofps.oreilly.com/titles/9781449303822/DialplanFundamentals.html 2011/10/27 Sammy Govind > It was a challenge to read through all the interesting experience you've
> shared over here. I don't know what others may be using for parsing the logs
> beautifully and make them usable. What I would recommend you…

Asterisk Users 3.9 years ago 0 Answers

OPTIONS support for SDP

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I have been sending OPTIONS requests 1) programatically (my own code),
2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes
in sip.conf. The trouble is I do not see anything except an ACK 200 come
back from endpoints and it does not contain any SDP/codec info. . My goal is
to determine audio and video codec capability in advance of a call INVITE. I
notice in both 2 and 3 examples the Asterisk generated OPTIONS does not
specify any ACCEPT header (ie Accept=application/sdp). I was thinking maybe
that is why I…

Asterisk Users 3.9 years ago 0 Answers

Unknown warning

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Hi Can anyone shed some light on what this warning means? chan_sip.c:19184 handle_response_invite: just did sched_add
waitid(1223301) for sip_reinvite_retry for dialog
3c46ab7f1762-8nxnhonpfcgr in handle_response_invite I've had a good look online but can't find a decent answer. Thanks in advance Ish

Asterisk Users 3.9 years ago 7 Answers

Still having trouble to configure gxw4108 with asterisk 1.8 need enlightenment

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Dear all,
I'm still having trouble using asterisk with the grandstream gxw4108,
in the gxw4108 I'm using 1 stage dialing in the profile1 I already
type my asterisk server address 192.168.14.80 and my grandstream IP is
192.168.101.184 here's my asterisk config files SIP.CONF
[1401]
type = friend
username = 1401
secret = 1401
host = dynamic
context = kantor-mtx
insecure = port
nat = yes
dtmfmode = rfc2833
canreinvite = yes
notifyringing = yes [1402]
type = friend
username = 1402
secret…

Asterisk Users 3.9 years ago 0 Answers

Tips & best practices for asterisk troubleshooting & parsing logs

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It was a challenge to read through all the interesting experience you've
shared over here. I don't know what others may be using for parsing the logs
beautifully and make them usable. What I would recommend you at the very
beginning ,since you mentioned using egrep, is figure out the Channel
identifier string from the logs for a particular call. That's underlined
below for you. [Oct 26 17:58:01] VERBOSE[14274] logger.c: -- Executing [s@tc-maint:3]
> System("*Local/s@tc-maint-2496,2*","/var/lib/asterisk/bin/schedtc.php 60
> /var/spool/asterisk/outgoing 0") in new stack
Once you Figure out this part use egrep tool…

Asterisk Users 3.9 years ago 0 Answers

Asterisk 1.8 RealTime problem with ipaddr field

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Dear Asterisk users,  
I use Asterisk for a long time with RealTime support. Until few days everything went alright. SIP registrations and call handling is still good,. but I saw a disturbing thing today when I was checking my RT database which is a MySQL database. I connect to MySQL with Asterisk 1.8 mysql module (so not with the ODBC driver).  
I saw a row where one of my extension's details are weren't complete. For example: the fullcontact filed was filled but the ipaddr filed filled with "(null)" value. All the other extensions are filled total,…

Asterisk Users 3.9 years ago 1 Answer

Tips & best practices for asterisk troubleshooting & parsing logs

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Hello all, I have been running asterisk systems since summer of 2008. I do not claim to be an expert. But I have worked through many issues during this period. I have setup & manage 5 systems, which serve 6 companies total (and of course process calls for all of the people they do business with). I have always been happy with asterisk (well, obviously less happy during the problem times... :-). And I continue to prefer to us it. However, if I could name the one largest struggle that I have with asterisk, it is the facilities that it…

Asterisk Users 3.9 years ago 0 Answers

OPTIONS to determine codec capability before an INVITE

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I have been sending OPTIONS requests 1) programatically (my own code),
2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes
in sip.conf. The trouble is I do not see anything except an ACK 200 come
back from endpoints and it does not contain any SDP/codec info. . My goal is
to determine audio and video codec capability in advance of a call INVITE. I
notice in both 2 and 3 examples the Asterisk generated OPTIONS does not
specify any ACCEPT header (ie Accept=application/sdp). I was thinking maybe
that is why I…

Asterisk Users 3.9 years ago 0 Answers