Hello every one, I’m new here,
Right now, I’m trying to set up a SIP Trunk between an Asterisk server and a P-CSCF(Open IMS Core)…But till now the it doesn’t working!
If there someone can help me please…I need a HELP.
thanks
there are plenty of people here who can try to help, however … what have you given us to work with ?
you need to explain what you’ve done in terms of i did X got Y but expected Z ; or something like that
sip debug is available within asterisk and you have tools like sngrep and wireshark
provide the steps you’ve taken to debug and then whats still failing at what level with details
Thanks for your quick reply!
Absolutly!..I’m biginer in this domain so I have a lot of problems…but in this case I have configured the Asterisk domain on a different machine than the IMS domain correctly and I can establish calls in each domain. But for the SIP Trunk between them I have the problem and I don’t know where is it.
Bellow there is my asterisk configuration
;########### config de serveur asterisk #############
[900]
username=900
type=friend
context=projet0
secret=900
host=dynamic
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=GSM
allow=g726
allow=h261
allow=h263
allow=h263p
qualify=yes
[911]
username=911
type=friend
context=projet0
secret=911
host=dynamic
nat=no
disallow=all
allow=ulaw
allow=alaw
allow=GSM
allow=g726
allow=h261
allow=h263
allow=h263p
qualify=yes
[asterisk_as]
username=asterisk_as
type=friend
insecure=port,invite
qualify=yes
canreinvite=no
host=192.168.1.10
context=projet0
extensions.conf
[projet0]
exten=>_9xx,1,Dial(SIP/${EXTEN})
exten=>100,1,Dial(SIP/alice@open-ims.test)
exten=>101,1,Dial(SIP/bob@open-ims.test)
Thanks again for helping me!
sip set debug on output?
You have copied a cookbook configuration without understanding it; I would advise going through the configuration against the documentation. Pending the trace output, I can see things that will compromise security or cause false authentication rejections, but nothing that will definitely break things.
Unfortunately, most of hte cook book solutions, on the web, are poor. They tend to be a decade out of date and do things that maximise the chance of working, at the expense of security.
first of all I will thank you.
In real I don’t know what shall I do about the SIP Trunk specially because in both sides I can make calls internally. By that I demand to you if you can that you explain to me what shall I do step by step.
thanks again!
Hi;
I’m installing asterisk manuelly on ubuntu 12.04 desktop and I have a problem:I can’t start asterisk CLI when I enter asterisk -rvvvvv I got this output: unable to connect to remote asterisk (does var/run/asterisk/asterisk.ctl exist?).
help please!
Look at the logs, to find out why the daemon is aborting. The symptom you describe is a generic symptom of a failure to complete startup.
what are the logs ? how can look at them?
But the problem is that I can not get the asterisk CLI to configure verbosity levels and rotate logs
Default logging levels can be set in asterisk.conf.
Log rotation is irrelevant for startup failures.
Hi,
I have configured a SIP Trunk between asterisk and open ims core. But when i start call between them I get this in the asterisk CLI
== Using SIP RTP CoS mark 5
– Executing [Alice@projet:1] Dial(“SIP/911-0000001c”, “SIP/alice@open-ims.test,20”) in new stack
== Using SIP RTP CoS mark 5
[Jul 28 07:34:39] ERROR[16053]: netsock2.c:263 ast_sockaddr_resolve: getaddrinfo(“open-ims.test”, “(null)”, …): No address associated with hostname
[Jul 28 07:34:39] WARNING[16053]: chan_sip.c:5483 create_addr: No such host: open-ims.test
[Jul 28 07:34:39] WARNING[16053]: app_dial.c:2218 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/911-0000001c’ status is ‘CHANUNAVAIL’
Can someone help me please!!
Thanks
What don’t you understand about the diagnostaic? It seem failly self evident.
Hi david551,
How it seem failly self evident?..and what do you suggest to solve?
This means that “open-ims.test” does not resolve to an IP address on the machine you are running Asterisk on. Since open-ims.test isn’t a real domain, you’ll either need to edit your /etc/asterisk/sip.conf file to put the actual IP address of the machine on it, or edit your /etc/hosts file to point that domain to your internal ip address. The “No address associated with hostname” line is your tipoff.
So, assuming your machine running open-ims.test has the ip address 10.1.1.5, you would look for a line in /etc/asterisk/sip.conf that looks like:
host=open-ims.test
and change it to
host=10.1.1.5 (or whatever the IP address is)
I can’t find "host=open-ims.test in “sip.conf”…and I have set the ip address of asterisk machine in /etc/hosts…But the problem still remain not resolved!!
It’s in extensions.conf according to your earlier postings:
exten=>100,1,Dial(SIP/alice@open-ims.test
/etc/hosts should work unless /etc/nsswitch.conf is unsuitably configured.
I set this also but the problem persists…what should I find in /etc/nsswitch file?
Use the man command to find out.
How to find out the /etc/nsswitch with the man command