Hello, in X-lite I just entered the Domain (IP address of HIPATH), user and password - that's all. I attached the pcap files of the freeswitch and the xlite sessions. It's really hard to find apropriate logging infos on the HIPATH :-( Dieter Am 29.09.2016 um 13:32 schrieb David Villasmil:
Can you post the sip authetication from xlite? As far as challenge is concerned, after FS sends the REGISTER, HIPATH is supposed to send a 401, to which FS should respond with the authentication details... but you're not even getting that 401.. that's why I asked you to look at xlite's config, make sure that you have the exact same realm in FS. Also post your xlite config (minus user/pass)
On Thu, Sep 29, 2016 at 1:27 PM, Peter Steinbach <lists@telefaks.de <mailto:lists@telefaks.de>> wrote:
Seems that Hipath does not know about the number 3999 or 3999@172.16.40.49 <mailto:3999@172.16.40.49>. Maybe it must be 3999@<hipath-domain_name>? Or maybe Hipath needs to have a challeange password in the initial invite? How do you register with your X-Lite?
Best regards Peter
On 09/29/16 12:53, Scholz, Dieter wrote:
Hello,
sorry for my stupid newbie questions. :-( I'm just starting with the whole VOIP stuff.
The redirect in the picture probably is caused by our internal routing. There's a more direct way from my workstation to the HIPATH (-1 hop). But this is the connection that works!!!
In the freeswitch part i do not see any redirects (the routing is different).
Do you think the HIPATH is terminating the connection or does Freeswitch not send the challenge? What do I need to adapt if Freeswitch sends incomplete data? Why can register with X-lite but not with Freeswitch? What could be the difference?
Dieter
Am 29.09.2016 um 12:40 schrieb David Villasmil:
You didn't say anything about redirects :( Also, you're not even receiving a challenge...
for redirects look at https://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect <https://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect>
Though i've never used redirects before with FS...
Sorry
On Thu, Sep 29, 2016 at 12:34 PM, Scholz, Dieter <rd-disc@gmx.net <mailto:rd-disc@gmx.net>> wrote:
Hello,
here is the output of ngrep: -------------------------------------------------- U 172.21.235.80:5080 <http://172.21.235.80:5080> -> 172.16.40.49:5060 <http://172.16.40.49:5060> REGISTER sip:172.16.40.49;transport=udp SIP/2.0. Via: SIP/2.0/UDP 172.21.235.80:5080;rport;branch=z9hG4bKvK9tmcy3Ue8vm. Max-Forwards: 70. From: <sip:3999@172.16.40.49 <mailto:sip%3A3999@172.16.40.49>>;tag=v3cSHNK1cUKtm. To: <sip:3999@172.16.40.49 <mailto:sip%3A3999@172.16.40.49>>. Call-ID: e78209da-144d-4c38-b88a-2d47564a9045. CSeq: 97236318 REGISTER. Contact: <sip:gw+rudolf@172.21.235.80 <mailto:sip%3Agw%2Brudolf@172.21.235.80>:5080;transport=udp;gw=rudolf>. Expires: 3600. User-Agent: FreeSWITCH-mod_sofia/1.6.10-17-726448d~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Content-Length: 0. .
# U 172.16.40.49:5060 <http://172.16.40.49:5060> -> 172.21.235.80:5080 <http://172.21.235.80:5080> SIP/2.0 100 Trying. Via: SIP/2.0/UDP 172.21.235.80:5080;rport=5080;branch=z9hG4bKvK9tmcy3Ue8vm;received=172.21.235.80. From: <sip:3999@172.16.40.49 <mailto:sip%3A3999@172.16.40.49>>;tag=v3cSHNK1cUKtm. To: <sip:3999@172.16.40.49 <mailto:sip%3A3999@172.16.40.49>>. Call-ID: e78209da-144d-4c38-b88a-2d47564a9045. CSeq: 97236318 REGISTER. Server: OpenScape 4000 - HiPath 4000 SoftGate. Content-Length: 0. .
# U 172.16.40.49:5060 <http://172.16.40.49:5060> -> 172.21.235.80:5080 <http://172.21.235.80:5080> SIP/2.0 404 Not Found. Via: SIP/2.0/UDP 172.21.235.80:5080;rport=5080;branch=z9hG4bKvK9tmcy3Ue8vm;received=172.21.235.80. From: <sip:3999@172.16.40.49 <mailto:sip%3A3999@172.16.40.49>>;tag=v3cSHNK1cUKtm. To: <sip:3999@172.16.40.49 <mailto:sip%3A3999@172.16.40.49>>;tag=160135440. Call-ID: e78209da-144d-4c38-b88a-2d47564a9045. CSeq: 97236318 REGISTER. Server: OpenScape 4000 - HiPath 4000 SoftGate. Content-Length: 0. --------------------------------------------------
I will see if I findd entries in the HIPATH log files (I'm not the admin of the HIPATH system). As realm I use the IP address of the HIPATH. I attached a picture with the relevant parts of the Wireshark log of the X-lite session.
Dieter
Am 29.09.2016 um 12:04 schrieb Peter Steinbach:
Can you post the SIP messages going to and from the Hipath? See my previous mail about ngrep.
On 09/29/16 11:50, Scholz, Dieter wrote:
Hello,
thanks for your answer.
I posted my sofia config in another mail today. I think that my config matches yours more or less. But when I start freeswitch the status of the connection is FAIL_WAIT. In the Wireshark log I saw that a REGISTER is send to the HIPATH but the answer is a 404. Any ideas what to test. Are there any sophia.conf.xml settings that are important?
Dieter
Am 29.09.2016 um 11:31 schrieb David Villasmil:
this is what my gateway look like:
<gateways> <gateway name="whatever"> <param name="username" value="1001"/> <param name="realm" value="THE REALM AGAINST UPON WHICH THE HASH WILL BE BUILT"/> <param name="password" value="password"/> <param name="extension" value="auto_to_user"/> <param name="context" value="public"/> <param name="proxy" value="IP ADDRESS OR DOMAIN"/> <param name="register" value="true"/> <param name="register-transport" value="tcp (or udp)"/> <param name="retry-seconds" value="15"/> </gateway> </gateways>
This should work... If you get Not Found, try looking at your server's log.
you also should create an outgoing gateway, even if you're not registering, like:
assuming the outgoing gateway will not auth-challenge you:
<gateway name="outgoing_gw"> <param name="username" value="none"/> <param name="realm" value="whatever"/> <param name="password" value="none"/> <param name="context" value="public"/> <param name="proxy" value="IP ADDRESS OR DOMAIN"/> <param name="register" value="false"/> <param name="register-transport" value="tcp (or udp)"/> <param name="retry-seconds" value="15"/> </gateway>
Adding it to the external is a good idea, so the final config will be:
<gateways>
<gateway name="whatever"> <param name="username" value="1001"/> <param name="realm" value="THE REALM AGAINST UPON WHICH THE HASH WILL BE BUILT"/> <param name="password" value="password"/> <param name="extension" value="auto_to_user"/> <param name="context" value="public"/> <param name="proxy" value="IP ADDRESS OR DOMAIN"/> <param name="register" value="true"/> <param name="register-transport" value="tcp (or udp)"/> <param name="retry-seconds" value="15"/> </gateway>
<gateway name="outgoing_gw"> <param name="username" value="does_not_matter_will_not_be used_but_must_be_set"/> <param name="realm" value="SAME AS PROXY"/> <param name="password" value="does_not_matter_will_not_be used_but_must_be_set"/> <param name="context" value="public"/> <param name="proxy" value="IP ADDRESS OR DOMAIN"/> <param name="register" value="false"/> <param name="register-transport" value="tcp (or udp)"/> <param name="retry-seconds" value="15"/> </gateway>
</gateways>
Once this is working and you see it REGED, calls will be coming in (as per "context") on the public dialplan... the you just need to bridge it like:
<extension name="forward call"> <condition field="destination_number" expression="^(whatever your regexp is)$"> <action application="info"/> <action application="bridge" data="sofia/gateway/outgoing_gw/$1"/> <action application="hangup"/> </condition> </extension>
I hope I was able to help..
Enjoy freeSWITCH, it's fantastic!
David
On Thu, Sep 29, 2016 at 9:40 AM, Scholz, Dieter <rd-disc@gmx.net <mailto:rd-disc@gmx.net> <mailto:rd-disc@gmx.net <mailto:rd-disc@gmx.net>>> wrote:
Hello,
thanks for your comment. The URL you mentioned will be helpful, when I reach the dialplan stage.
But at the moment I have problems connecting to the HIPATH maschine. I gave some more details in another post.
Dieter
Am 28.09.2016 um 17:25 schrieb Stanislav Sinyagin: > you need to create one more gateway that will communicate to your SIP > provider. This gateway can be configured with username and password if > the ITSP requires authentication. > > Then, you will need to create the dialplan that accepts calls from > each of these gateways and routes them to the other gateway. > > It's quite trivial once you figure out how SIP profiles and dialplan > contexts work together. > > Hopefully this will provide you some hints: > https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+PBX+Example <https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+PBX+Example>
<https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+PBX+Example <https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+PBX+Example>> > > > > > On Wed, Sep 28, 2016 at 2:22 PM, Scholz, Dieter <rd-disc@gmx.net <mailto:rd-disc@gmx.net> <mailto:rd-disc@gmx.net <mailto:rd-disc@gmx.net>>> wrote: >> Hello, >> >> this is my problem: >> >> I would like to use Freeswitch as a gateway between our Hipath PBX and >> an external SIP provider. >> >> For that I created a Hipath SIP account that is working (tested with >> X-lite). Now I would like to let Freeswitch act as a client using this >> SIP account. Whenever there is an incoming call Freeswitch should act on >> it and forward this call to an external SIP provider. >> >> Is this possible? Does that make sense? Are there alternatives? >> >> At the moment I'm stuck to configure Freeswitch as a client. I added the >> Hipath SIP account as a gateway. When I compare the SIP traffic of >> X-lite and Freeswitch using Wireshark I found out that Freeswitch dows >> not offer the 'Subscribe' option. Is this the reason why it fails? >> >> Can you help me? >> >> Thanks in advance. >> >> Dieter >> >>
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