[HT-503] - Can't do calls PSTN to VOIP

ip-communications

#1

We use an integrated HT-503 to asterisk, I’m having a problem where I can’t originate calls PSTN to VOIP. Calls originating from the VOIP to the telephone system (VOIP to PSTN) occur normally.

Log in a call:

Jul 19 17:06:19 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] SIPStack(1)::run: Active transactions: 6
Jul 19 17:06:24 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] FXO_POLARITY event received on port 1 ch 0
Jul 19 17:06:24 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] EventManager::run: Dispatching event 31 (LINE_POLARITY_CHANGED) on port 1:0
Jul 19 17:06:34 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] SIPStack(1)::run: Active transactions: 5
Jul 19 17:06:45 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] SIPStack(1)::isMessegeFromAllowedProxy, acct= 0, server= NULL:5060, failover= NULL:5060, outboundproxy= NULL:5060
Jul 19 17:06:45 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] SIPStack(1)::isMessegeFromAllowedProxy, acct= 1, server= [IP-VOIP-SERVER]:5060, failover= NULL:5060, outboundproxy= NULL:5060
Jul 19 17:06:45 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] SIPStack(1)::parseMessage: Mark server [IP-VOIP-SERVER] available
Jul 19 17:06:45 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] SIPStack(1)::cb_rcvreq: Received SIP request OPTIONS
Jul 19 17:06:45 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] SIPStack(1)::run: Active transactions: 5
Jul 19 17:07:01 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] FXO_POLARITY event received on port 1 ch 0
Jul 19 17:07:01 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] EventManager::run: Dispatching event 31 (LINE_POLARITY_CHANGED) on port 1:0
Jul 19 17:07:01 192.168.2.160 HT-503 [00:0B:XX:XX:XX:XX]: [1.0.15.5] SIPStack(1)::run: Active transactions: 4

I upgraded the firmware to 1.0.16.2, but it did not help.
Sorry for my google translate english.
If anyone can help me, thank you in advance.


#2

Did you set unconditional forward to voip on Basic page ?


#3

Yes, it worked for a long time:


#4

Do you see incoming packets on server ?


#5

Sorry, packet on server:

11:36:47.267962 IP (tos 0x0, ttl 56, id 22654, offset 0, flags [none], proto UDP (17), length 524)
[IP-CALLER] > [IP-VOIP-SERVER]: [udp sum ok] SIP, length: 496
SIP/2.0 200 OK
Via: SIP/2.0/UDP [IP-VOIP-SERVER]:5060;branch=z9hG4bK25606459;rport=5060
From: “Unknown” sip:Unknown@IP-VOIP-SERVER;tag=as5ea3d83e
To: sip:[TRUNK-NAME]@IP-FXO:5062;tag=1213371245
Call-ID: 15f5b86e0bb6aec159ff33c941d503c4@[IP-VOIP-SERVER]:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V2.0A 1.0.15.5 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0


#6

I just receive this and REGISTER packets, in PSTN to VOIP calls I don’t receive any packet in VOIP server.
The HT503 are logged in, but don’t are “forwarding call to server”.


#7

Then it must be incoming call is not detected by HT, check if PSTN provider change something on line.
Also try simply end this line with analogue phone, then if works try FXS port yo receive call.