Hi all,
Hope someone can help me with this.
The main objective are to make Asterisk and SER communicate with each other. Call SER--> Asterisk and Asterisk--> SER.
I used the example configuration for pstn as base on all the docs, it is how we can make ser and asterisk work. As instructed on the docs, we just need to add the IP address of the PSTN gateway on the trusted database of our SER, in this case, asterisk is our PSTN gateway.
=================== mysql> insert into trusted values ("192.168.1.247","any","^sip:.*$"); ===================
On the asterisk (trixbox) server, below are the configuration I defined:
Outgoing Setting: Trunk Name: serout Peer Details: ==================== allow=all dtmfmode=rfc2833 host=192.168.1.41 insecure=no type=peer ====================
I can ring any number on my SER Server using any number on my Asterisk, problem is when I pick-up the phone I cannot hear voice and according to the logs as listed below, it is hanging-up and something like "Unresolvable destination".
SER NGREP RESULT #ngrep -n 5060 -d eth0 3242194ngrep -n 5060 -d eth0 3242194 ================================= U 192.168.1.41:5060 -> 192.168.1.247:5060 SIP/2.0 478 Unresolvable destination (478/TM)..Via: SIP/2.0/UDP 192.168.1.247:5060;branch=z9hG4bK580c7aec;rport=5060..F rom: "3000" sip:3000@192.168.1.247;tag=as7693144e..To: sip:3242194@192.168.1.41;tag=419e8..Call-ID: 5784a2681edd513 c59c77e20512b499d@192.168.1.247..CSeq: 103 INVITE..Server: Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0.. Warning: 392 192.168.1.41:5060 "Noisy feedback tells: pid=9413 req_src_ip=192.168.1.247 req_src_port=5060 in_uri=sip:3 242194@192.168.1.6:52961;user=phone out_uri=sip:3242194@192.168.1.6:52961;user=phone via_cnt==1".... =================================
Asterisk Logs ================================= -- Executing NoOp("SIP/3000-08fd48b8", "CallerID set to "3000" <3000>") in new stack -- Executing Set("SIP/3000-08fd48b8", "GROUP()=OUT_2") in new stack -- Executing GotoIf("SIP/3000-08fd48b8", "0?108") in new stack -- Executing Set("SIP/3000-08fd48b8", "DIAL_NUMBER=3242194") in new stack -- Executing Set("SIP/3000-08fd48b8", "DIAL_TRUNK=2") in new stack -- Executing AGI("SIP/3000-08fd48b8", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/3000-08fd48b8", "OUTNUM=3242194") in new stack -- Executing Set("SIP/3000-08fd48b8", "custom=SIP/Serout") in new stack -- Executing GotoIf("SIP/3000-08fd48b8", "0?16") in new stack -- Executing Dial("SIP/3000-08fd48b8", "SIP/Serout/3242194|120|r") in new stack -- Called Serout/3242194 -- SIP/Serout-08fda208 is ringing -- SIP/Serout-08fda208 answered SIP/3000-08fd48b8 -- Attempting native bridge of SIP/3000-08fd48b8 and SIP/Serout-08fda208 -- Got SIP response 478 "Unresolvable destination (478/TM)" back from 192.168.1.41 == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/3000-08fd48b8' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/3000-08fd48b8'
=================================
Can anyone help me resolve this problem.
Thanks in advanced.
Rjey
____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/
This means you can't resolve 192.168.1.247 from your SER server. Try adding an entry for it into /etc/hosts to see if you can bypass DNS.
N.
Rjey Nomer wrote:
Hi all,
Hope someone can help me with this.
The main objective are to make Asterisk and SER communicate with each other. Call SER--> Asterisk and Asterisk--> SER.
I used the example configuration for pstn as base on all the docs, it is how we can make ser and asterisk work. As instructed on the docs, we just need to add the IP address of the PSTN gateway on the trusted database of our SER, in this case, asterisk is our PSTN gateway.
=================== mysql> insert into trusted values ("192.168.1.247","any","^sip:.*$"); ===================
On the asterisk (trixbox) server, below are the configuration I defined:
Outgoing Setting: Trunk Name: serout Peer Details: ==================== allow=all dtmfmode=rfc2833 host=192.168.1.41 insecure=no type=peer ====================
I can ring any number on my SER Server using any number on my Asterisk, problem is when I pick-up the phone I cannot hear voice and according to the logs as listed below, it is hanging-up and something like "Unresolvable destination".
SER NGREP RESULT #ngrep -n 5060 -d eth0 3242194ngrep -n 5060 -d eth0 3242194 ================================= U 192.168.1.41:5060 -> 192.168.1.247:5060 SIP/2.0 478 Unresolvable destination (478/TM)..Via: SIP/2.0/UDP 192.168.1.247:5060;branch=z9hG4bK580c7aec;rport=5060..F rom: "3000" sip:3000@192.168.1.247;tag=as7693144e..To: sip:3242194@192.168.1.41;tag=419e8..Call-ID: 5784a2681edd513 c59c77e20512b499d@192.168.1.247..CSeq: 103 INVITE..Server: Sip EXpress router (0.9.3 (i386/linux))..Content-Length: 0.. Warning: 392 192.168.1.41:5060 "Noisy feedback tells: pid=9413 req_src_ip=192.168.1.247 req_src_port=5060 in_uri=sip:3 242194@192.168.1.6:52961;user=phone out_uri=sip:3242194@192.168.1.6:52961;user=phone via_cnt==1".... =================================
Asterisk Logs
-- Executing NoOp("SIP/3000-08fd48b8", "CallerID set to "3000" <3000>") in new stack -- Executing Set("SIP/3000-08fd48b8", "GROUP()=OUT_2") in new stack -- Executing GotoIf("SIP/3000-08fd48b8", "0?108") in new stack -- Executing Set("SIP/3000-08fd48b8", "DIAL_NUMBER=3242194") in new stack -- Executing Set("SIP/3000-08fd48b8", "DIAL_TRUNK=2") in new stack -- Executing AGI("SIP/3000-08fd48b8", "fixlocalprefix") in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Set("SIP/3000-08fd48b8", "OUTNUM=3242194") in new stack -- Executing Set("SIP/3000-08fd48b8", "custom=SIP/Serout") in new stack -- Executing GotoIf("SIP/3000-08fd48b8", "0?16") in new stack -- Executing Dial("SIP/3000-08fd48b8", "SIP/Serout/3242194|120|r") in new stack -- Called Serout/3242194 -- SIP/Serout-08fda208 is ringing -- SIP/Serout-08fda208 answered SIP/3000-08fd48b8 -- Attempting native bridge of SIP/3000-08fd48b8 and SIP/Serout-08fda208 -- Got SIP response 478 "Unresolvable destination (478/TM)" back from 192.168.1.41 == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/3000-08fd48b8' in macro 'dialout-trunk' == Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on 'SIP/3000-08fd48b8'
=================================
Can anyone help me resolve this problem.
Thanks in advanced.
Rjey
Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase. http://farechase.yahoo.com/ _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
Rjey Nomer -
SIP