problem with internal calls (sip to sip)

fabianus

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#1
Hello guys !

Strange problem that we have :

Incomming calls from a sip-trunk work fine. Outgoing calls from a sip-trunk work fine to.

BUT calls from one sip account to anothere inside of Asterisk don't come through !
Error message : "Call failed, declined" or "not reachable" depending on the softphone.
On the reports page the call is marked "NO ANSWER".
It seems that Asterisk rejects the call.

Thanks a lot for any help !

Regards,
Fabianus

Post edited by: fabianus, at: 2008/03/22 03:19<br><br>Post edited by: fabianus, at: 2008/03/22 04:28
 

rejil.rajan

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#2
Hi

Can you give the command sip show peers and see if the SIP phones are coming and registering to the Elastix server. Confirm the IPs of the phone with that on the screen after giving the command sip show peers. If no IP address is shown after giving this command, then it means that the configuration on the phone is wrong

Thanks & Regards
 

fabianus

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#3
Hello Rejil Rajan,

Thanks for your kind support.

Here is the report :

Name/username Host Dyn Nat ACL Port Status
33175439999/33175439999 83.99.99.72 5060 OK (90 ms)
2112/2112 192.168.2.43 D N 28936 OK (103 ms)
2111 (Unspecified) D N 0 UNKNOWN
2110 (Unspecified) D N 0 UNKNOWN
2109 (Unspecified) D N 0 UNKNOWN
2108 (Unspecified) D N 0 UNKNOWN
2107 (Unspecified) D N 0 UNKNOWN
2106 (Unspecified) D N 0 UNKNOWN
2105 (Unspecified) D N 0 UNKNOWN
2104/2104 192.168.2.22 D N 5060 OK (57 ms)
2103/2103 192.168.2.22 D N 5060 OK (58 ms)
2102/2102 192.168.2.49 D N 25948 OK (102 ms)
2101/2101 192.168.2.34 D N 5060 OK (96 ms)
2100/2100 192.168.2.34 D N 5060 OK (96 ms)
14 sip peers [Monitored: 7 online, 7 offline Unmonitored: 0 online, 0 offline]


I wanna call for example from 2100 to 2101, but it doesn't call ...

Thanks for any further suggestion !

Regards,
Fabianus
 

rejil.rajan

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#4
Hi Fabinus

As i can see from those logs

2101/2101 192.168.2.34 D N 5060 OK (96 ms)
2100/2100 192.168.2.34 D N 5060 OK (96 ms)

Both 2101 and 2100 are having the same IP address. Have they been configured on the same phone. Are you able to make a call from 2100 to 2112.

Cos from the same device your being registered on the same port which is wrong, U should be having 2100 on 5060 and 2101 on 5061 or vice versa
 

mbit

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#5
Did you upgrade to FreePBX 2.4?
 

lek

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#6
An asterisk full log could help. Plase send the log with verbose 9 or higher.
 

fabianus

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#7
Hello all !

Excuse-me for not responding for so long.

In fact I stoped the Asterisk project for some time, and now that I am back I run into the same problem.

I installed Elastix 1.0-15 and sip show peers gives me this :

Name/username Host Dyn Nat ACL Port Status
2910/2910 164.0.1.74 D N 5260 OK (8 ms)
2902 (Unspecified) D N 0 UNKNOWN
2817 (Unspecified) D N 0 UNKNOWN
2805 (Unspecified) D N 0 UNKNOWN
2729/2729 164.0.1.71 D N 5161 OK (9 ms)
2728/2728 164.0.1.71 D N 5261 OK (8 ms)
2727 (Unspecified) D N 0 UNKNOWN
2726/2726 164.0.1.71 D N 5061 OK (8 ms)
2725/2725 164.0.1.72 D N 5160 OK (8 ms)
2724 (Unspecified) D N 0 UNKNOWN
2723 (Unspecified) D N 0 UNKNOWN
2722 (Unspecified) D N 0 UNKNOWN
2721/2721 164.0.1.71 D N 5160 OK (8 ms)
2719/2719 164.0.1.72 D N 5061 OK (8 ms)
2713 (Unspecified) D N 0 UNKNOWN
2712 (Unspecified) D N 0 UNKNOWN
2711/2711 164.0.1.71 D N 5260 OK (8 ms)
2600/2600 164.0.1.74 D N 5360 OK (7 ms)
2302/2302 164.0.1.74 D N 5361 OK (8 ms)
2301/2301 164.0.1.71 D N 5060 OK (8 ms)
2138 (Unspecified) D N 0 UNKNOWN
2137 (Unspecified) D N 0 UNKNOWN
2136/2136 164.0.1.73 D N 5061 OK (8 ms)
2130/2130 164.0.1.73 D N 5160 OK (8 ms)
2128/2128 164.0.1.74 D N 5160 OK (9 ms)
2125 (Unspecified) D N 0 UNKNOWN
2119/2119 164.0.1.74 D N 5261 OK (9 ms)
2108 (Unspecified) D N 0 UNKNOWN
2107/2107 164.0.1.182 D N 38919 OK (104 ms)
2105/2105 164.0.1.73 D N 5161 OK (8 ms)
2102/2102 164.0.1.73 D N 5260 OK (8 ms)
2101 (Unspecified) D N 0 UNKNOWN
2100/2100 164.0.1.72 D N 5060 OK (9 ms)
2099/2099 164.0.1.74 D N 5061 OK (9 ms)
2086 (Unspecified) D N 0 UNKNOWN
2084/2084 164.0.1.74 D N 5060 OK (9 ms)
36 sip peers [Monitored: 21 online, 15 offline Unmonitored: 0 online, 0 offline]


I am still not able to make calls from one sip to another internal sip account.

Thanks a lot for any feedback !

Regards,
Fabianus
 

fabianus

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#8
Hello,

finally I set in the sip.conf context = from-trunk
and this fixed it.

BUT I now have a diffrent problem : when I call an internal sip account from another internal sip account, once I dialed it takes 10 sec. to start to ring (in between there is nothing).

Any idea from what this could come from?

Thanks a lot for any feedback !

Regards,
Fabianus
 

adminad

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#9
Are you using SIP/IAX trunks? How many?
 

fabianus

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#10
Hi Admin,

thanks for your interest!

I use sip accounts, you have the list above (about 38).

I just did a new post on this problem, as it is very diffrent from this one. Up to you where you would like to continue to discuss.
http://www.elastix.org/index.php?option ... ith+1.0-15

Thanks,
Fabianus
 

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