GoiP trunk to GoIP trunk calls hang ups

Good day everyone!

I have a problem with two GSM trunks to bridge a call. I currently have two GoIP boxes connected to my server. When a call is placed on GoiP A, a number is dialed to GoiP B in order to forward it to a specific mobile number. The call happens and the other mobile number rings. The problem is, when the number being called to answers, GoIP A hangs up the call. Is there any config that I’m missing?

Thanks in Advance for your help.

How could we possibly know? You did not tell us your config or provide a log excerpt from when the call drops.

Hi Skyking,

My two goips and placed on the same LAN network. Config is shown below.

username=gsm1
fromuser=gsm1
secret=mypass
context=goip
host=dynamic
port=5061
type=friend
canreinvite=no
nat=no
qualify=yes
insecure=very

Im using asterisk 10.7 and freepbx 2.10.

Logs is shown below

[2012-08-31 15:42:17] VERBOSE[8103] netsock2.c: == Using SIP RTP TOS bits 184
[2012-08-31 15:42:17] VERBOSE[8103] netsock2.c: == Using SIP RTP CoS mark 5
[2012-08-31 15:42:17] VERBOSE[8103] app_dial.c: – Called SIP/gsm2/0445369851
[2012-08-31 15:42:17] VERBOSE[8103] app_dial.c: – SIP/gsm2-00000014 is making progress passing it to SIP/gsm1-00000013
[2012-08-31 15:42:17] VERBOSE[8103] app_dial.c: – SIP/gsm2-00000014 is making progress passing it to SIP/gsm1-00000013
[2012-08-31 15:42:29] VERBOSE[8103] app_dial.c: – SIP/gsm2-00000014 answered SIP/gsm1-00000013
[2012-08-31 15:42:35] VERBOSE[8103] pbx.c: – Executing [[email protected]:1] Macro(“SIP/gsm1-00000013”, “hangupcall,”) in new stack
[2012-08-31 15:42:35] VERBOSE[8103] pbx.c: – Executing [[email protected]:1] GotoIf(“SIP/gsm1-00000013”, “1?theend”) in new stack
[2012-08-31 15:42:35] VERBOSE[8103] pbx.c: – Goto (macro-hangupcall,s,3)
[2012-08-31 15:42:35] VERBOSE[8103] pbx.c: – Executing [[email protected]:3] ExecIf(“SIP/gsm1-00000013”, “0?Set(CDR(recordingfile)=)”) in new stack
[2012-08-31 15:42:35] VERBOSE[8103] pbx.c: – Executing [[email protected]:4] Hangup(“SIP/gsm1-00000013”, “”) in new stack
[2012-08-31 15:42:35] VERBOSE[8103] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/gsm1-00000013’ in macro ‘hangupcall’
[2012-08-31 15:42:35] VERBOSE[8103] features.c: == Spawn extension (macro-dial-one, h, 1) exited non-zero on ‘SIP/gsm1-00000013’
[2012-08-31 15:42:35] VERBOSE[8103] app_macro.c: == Spawn extension (macro-dial-one, s, 42) exited non-zero on ‘SIP/gsm1-00000013’ in macro ‘dial-one’

Above logs shows that somehow the calls are being relayed.

Just found out that it takes 4 rings to pass the call in between the 2 GoIP modems and the network allows around 8 rings only, on my test this only gives the target mobile 2 vibrates and it disconnects the call, otherwise the call will proceed if answered right away. Any body knows how to lessen the delay for diverts inside GoiP?

regards

I don’t know anything about the GoIP but it sounds like you don’t have a dial plan loaded so you have to wait for the digit timeout.

Does it have a local log or support syslog to watch the call progress in real time?

I can’t see any local logs on this device. I try to see about the dialplan for this device and get back to the forum…

Can’t see any setting on this device to reduce the delays passing the calls from one area to another. The best way I found was to place an IVR in between that answers the call thus preventing the ring timeout from the line and giving the caller more options.

If you have made an open-ended dialplan, try terminating your dialstring with a # until you fix it.