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Problem with IVR/RING GROUP SIP+Retransmissions or Nativ Bridge?


(Dragan) #1

Hello, I have one problem that I can not solve :confused:
I use FreePBX 13.0.192.16
I’m connected to a sip trunk to the provider who uses it BroadSoft IMS
Most things work properly except for one situation…when the incoming call goes to the IVR…On option 1, call on RING GROUP…the call is switched to the selected RING GROUP and ends after 2 ringing :confused:
If I look at ASTERISK log I get the following reason for the break:
[2019-01-24 10:11:26] WARNING[2176]: chan_sip.c:4038 retrans_pkt: Retransmission timeout reached on transmission asbcBW101113285240119-1467802376@10.10.10.4 for seqno 103 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 6400ms with no response
[2019-01-24 10:11:26] WARNING[2176]: chan_sip.c:4067 retrans_pkt: Hanging up call asbcBW101113285240119-1467802376@10.10.10.4 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
The first thing I thought was the problem is SIP+Retransmissions…since the address 10.10.10.4 is not available with ASTERISK.
After pressing the button 1 on the IVR when I look at the SIP packet header, I see the Call-ID field:
Call-ID: asbcBW101113285240119-1467802376@10.10.10.4
It was logical to me that if the provider allows me to communicate with 10.10.10.4 that it will solve the problem
I contacted my service provider about the problem…they say that there are 10.10.10.4 there, but I do not have to communicate with…as if he was just creating an Call-ID or something like that.We made debug calls on both sides…Here’s what ASTERISK says:SIP Debug
The call was made from the number 057931004–>Call number:+38757226180, On option 1, call on RING GROUP 288 (in which the extension 222 is currently located) and ends after 2 ringing :confused:

After reviewing the call, the ISP claims to be a problem NATIV BRIDGE…and says that after pressing the key 1, ASTERISK should not send any signaling to the ISP. The suggestion was to add the:canreinvite=no
I tried but the same condition :confused:
here is my current OUTGOING peer:
username=+38757201650@mtel.ba
type=peer
trustrpid=yes
sendrpid=pai
secret=xxxxxx
qualify=yes
insecure=port,invite
host=mtel.ba
fromuser=+38757201650
outboundproxy=10.252.64.110
fromdomain=mtel.ba
disallow=all
allow=alaw&ulaw&gsm&g729
context=custom-get-did-from-sip
dtmfmode=auto
canreinvite=no
nat=auto
And after adding the canreinvite=no line, the provider says it gets signalized after pressing the key 1 on the IVR.
I would also note that I have achieved SIPTRANK with another provider and that through this line everything is working properly regarding the same situation as this.
Have I ever been on the right way to solve the problem? Does anyone have an idea of what could be a problem? I currently have no idea what else I could try :confused:


(Dragan) #2

To answer to myself :grinning: Instead of RING GROUP I use QUEUES, and everything works right. I did not understand why the real problem with RING GROUP :face_with_monocle: