I already define the hostname of both machines and I created also an internal DNS wherein I define the following:
================================== # dig -t SRV _sip._udp.rjey.ph
; <<>> DiG 9.2.4 <<>> -t SRV _sip._udp.rjey.ph ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 13180 ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 3, AUTHORITY: 2, ADDITIONAL: 5
;; QUESTION SECTION: ;_sip._udp.rjey.ph. IN SRV
;; ANSWER SECTION: _sip._udp.rjey.ph. 10800 IN SRV 0 0 5060 asterisk.rjey.ph. _sip._udp.rjey.ph. 10800 IN SRV 0 0 5060 ser.rjey.ph.
;; AUTHORITY SECTION: rjey.ph. 10800 IN NS ns1.rjey.ph. rjey.ph. 10800 IN NS ns2.rjey.ph.
;; ADDITIONAL SECTION: ser.rjey.ph. 10800 IN A 192.168.1.41 asterisk.rjey.ph. 10800 IN A 192.168.1.247 ns1.rjey.ph. 10800 IN A 192.168.1.7
;; Query time: 46 msec ;; SERVER: 192.168.1.7#53(192.168.1.7) ;; WHEN: Tue Jun 5 18:32:58 2007 ;; MSG SIZE rcvd: 292 ============================
What else should I do????
Thanks,
Rjey
Date: Tue, 05 Jun 2007 07:24:27 -0400
From: "SIP" sip@arcdiv.com Add to Address Book Add Mobile Alert
To: "Rjey Nomer" rjeynomer@yahoo.com
CC: serusers@iptel.org
Subject: Re: [Serusers] SER and Asterisk
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/3000-08fd48b8and 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
____________________________________________________________________________________ Don't get soaked. Take a quick peak at the forecast with the Yahoo! Search weather shortcut. http://tools.search.yahoo.com/shortcuts/#loc_weather
Can you try these things for Asterisk: 1. nat=yes 2. insecure=very (just try and see if this works) 3. Since you are using Asterisk, try to set Allow anonymous SIP calls to "yes" in the general settings of FreePBX interface
On 6/5/07, Rjey Nomer rjeynomer@yahoo.com wrote:
I already define the hostname of both machines and I created also an internal DNS wherein I define the following:
================================== # dig -t SRV _sip._udp.rjey.ph
; <<>> DiG 9.2.4 <<>> -t SRV _sip._udp.rjey.ph ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 13180 ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 3, AUTHORITY: 2, ADDITIONAL: 5
;; QUESTION SECTION: ;_sip._udp.rjey.ph. IN SRV
;; ANSWER SECTION: _sip._udp.rjey.ph. 10800 IN SRV 0 0 5060 asterisk.rjey.ph. _sip._udp.rjey.ph. 10800 IN SRV 0 0 5060 ser.rjey.ph.
;; AUTHORITY SECTION: rjey.ph. 10800 IN NS ns1.rjey.ph. rjey.ph. 10800 IN NS ns2.rjey.ph.
;; ADDITIONAL SECTION: ser.rjey.ph. 10800 IN A 192.168.1.41 asterisk.rjey.ph. 10800 IN A 192.168.1.247 ns1.rjey.ph. 10800 IN A 192.168.1.7
;; Query time: 46 msec ;; SERVER: 192.168.1.7#53(192.168.1.7) ;; WHEN: Tue Jun 5 18:32:58 2007 ;; MSG SIZE rcvd: 292 ============================
What else should I do????
Thanks,
Rjey
Date: Tue, 05 Jun 2007 07:24:27 -0400
From: "SIP" sip@arcdiv.com Add to Address Book Add Mobile Alert
To: "Rjey Nomer" rjeynomer@yahoo.com
CC: serusers@iptel.org
Subject: Re: [Serusers] SER and Asterisk
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/3000-08fd48b8and 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
Don't get soaked. Take a quick peak at the forecast with the Yahoo! Search weather shortcut. http://tools.search.yahoo.com/shortcuts/#loc_weather _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
In no.3 i meant Trixbox, not Asterisk :)
On 6/5/07, Andrey Kuprianov andrey.kouprianov@gmail.com wrote:
Can you try these things for Asterisk:
- nat=yes
- insecure=very (just try and see if this works)
- Since you are using Asterisk, try to set Allow anonymous SIP calls
to "yes" in the general settings of FreePBX interface
On 6/5/07, Rjey Nomer rjeynomer@yahoo.com wrote:
I already define the hostname of both machines and I created also an internal DNS wherein I define the following:
================================== # dig -t SRV _sip._udp.rjey.ph
; <<>> DiG 9.2.4 <<>> -t SRV _sip._udp.rjey.ph ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 13180 ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 3, AUTHORITY: 2, ADDITIONAL: 5
;; QUESTION SECTION: ;_sip._udp.rjey.ph. IN SRV
;; ANSWER SECTION: _sip._udp.rjey.ph. 10800 IN SRV 0 0 5060 asterisk.rjey.ph. _sip._udp.rjey.ph. 10800 IN SRV 0 0 5060 ser.rjey.ph.
;; AUTHORITY SECTION: rjey.ph. 10800 IN NS ns1.rjey.ph. rjey.ph. 10800 IN NS ns2.rjey.ph.
;; ADDITIONAL SECTION: ser.rjey.ph. 10800 IN A 192.168.1.41 asterisk.rjey.ph. 10800 IN A 192.168.1.247 ns1.rjey.ph. 10800 IN A 192.168.1.7
;; Query time: 46 msec ;; SERVER: 192.168.1.7#53(192.168.1.7) ;; WHEN: Tue Jun 5 18:32:58 2007 ;; MSG SIZE rcvd: 292 ============================
What else should I do????
Thanks,
Rjey
Date: Tue, 05 Jun 2007 07:24:27 -0400
From: "SIP" sip@arcdiv.com Add to Address Book Add Mobile Alert
To: "Rjey Nomer" rjeynomer@yahoo.com
CC: serusers@iptel.org
Subject: Re: [Serusers] SER and Asterisk
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/3000-08fd48b8and 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
Don't get soaked. Take a quick peak at the forecast with the Yahoo! Search weather shortcut. http://tools.search.yahoo.com/shortcuts/#loc_weather _______________________________________________ Serusers mailing list Serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-
Andrey Kuprianov -
Rjey Nomer