Hello I've tryed to use netvoip.ch as a gateway. But I have a lot of connection problems. Because netvoip has a load balancing between two servers sip.netvoip.ch (sip-1.netvoip.ch and sip-2.netvoip.ch). Now the problem is if you are getting with the INVITE to the first server you're getting a 304 Unauthorized with the nonce of the first server. For some reason FreeSWITCH is now sending a INVITE with the digist Authentication to the second server and this server is now answering with 304 Unauthorized with an other nonce. FreeSWITCH now ends the call with 500 internal server error. Is this a FreeSWITCH related issue or a netvoip related issue? Or is there any configuration in the gateway to say to use the same server for the authentication process? Regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-tp75... Sent from the freeswitch-users mailing list archive at Nabble.com.
Not sure if it's the same problem, but it sounds similar to the issues we had a couple years ago getting FreeSwitch to work with Whistlephone. Whistlephone also used multiple servers for load balancing, and while outgoing calls worked fine, incoming calls were unreliable. For more technical details, you can search this mailing list for Whistlephone, about a year and a half ago. The solution was to add a separate gateway for each proxy server. So, in conf/sip_profiles/external, I added whistlephone.xml, with the stuff below. Pay attention to proxy vs proxy1 in the server names, and the added "from-domain" in the secondary gateways. Using the "realm" parameter seemed to help me, but it took experimentation to find what worked. -------------------------------------------------------------------- <code> <include> <gateway name="WhistlePhone"> <param name="username" value="xxxxxxxxxxx"/> <param name="from-user" value="xxxxxxxxxx"/> <param name="proxy" value="proxy.WhistlePhone.com"/> <param name="realm" value="WhistlePhone"/> <param name="password" value="xxxx"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway> <gateway name="WhistlePhoneProxy1"> <param name="username" value="xxxxxxxxxxx"/> <param name="from-user" value="xxxxxxxxxx"/> <param name="proxy" value="proxy1.WhistlePhone.com"/> <param name="from-domain" value="proxy.Whistlephone.com"/> <param name="password" value="xxxx"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway> repeat for each proxy server..... </include> </code> ------------------------------------------------------------------------
Hi Tom Thanks for your answer, but we have the problem with the outgoing calls. See attached sip trace. regards Patrick On Mon, Jun 11, 2012 at 1:09 AM, Tom C <mel0torme@gmail.com> wrote:
Not sure if it's the same problem, but it sounds similar to the issues we had a couple years ago getting FreeSwitch to work with Whistlephone. Whistlephone also used multiple servers for load balancing, and while outgoing calls worked fine, incoming calls were unreliable.
For more technical details, you can search this mailing list for Whistlephone, about a year and a half ago.
The solution was to add a separate gateway for each proxy server.
So, in conf/sip_profiles/external, I added whistlephone.xml, with the stuff below. Pay attention to proxy vs proxy1 in the server names, and the added "from-domain" in the secondary gateways. Using the "realm" parameter seemed to help me, but it took experimentation to find what worked.
-------------------------------------------------------------------- <code>
<include> <gateway name="WhistlePhone"> <param name="username" value="xxxxxxxxxxx"/> <param name="from-user" value="xxxxxxxxxx"/> <param name="proxy" value="proxy.WhistlePhone.com"/> <param name="realm" value="WhistlePhone"/> <param name="password" value="xxxx"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway>
<gateway name="WhistlePhoneProxy1"> <param name="username" value="xxxxxxxxxxx"/> <param name="from-user" value="xxxxxxxxxx"/> <param name="proxy" value="proxy1.WhistlePhone.com"/> <param name="from-domain" value="proxy.Whistlephone.com"/> <param name="password" value="xxxx"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway>
repeat for each proxy server.....
</include>
</code> ------------------------------------------------------------------------
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Hello I just got some information form the Provider, they use DNS for Loadbalancing, is there a way to tell Freeswitch to don't lookup agin. regards Patrick On Mon, Jun 11, 2012 at 8:08 AM, Patrick Dürsteler <patrick@sunsus.net>wrote:
Hi Tom
Thanks for your answer, but we have the problem with the outgoing calls. See attached sip trace.
regards
Patrick
On Mon, Jun 11, 2012 at 1:09 AM, Tom C <mel0torme@gmail.com> wrote:
Not sure if it's the same problem, but it sounds similar to the issues we had a couple years ago getting FreeSwitch to work with Whistlephone. Whistlephone also used multiple servers for load balancing, and while outgoing calls worked fine, incoming calls were unreliable.
For more technical details, you can search this mailing list for Whistlephone, about a year and a half ago.
The solution was to add a separate gateway for each proxy server.
So, in conf/sip_profiles/external, I added whistlephone.xml, with the stuff below. Pay attention to proxy vs proxy1 in the server names, and the added "from-domain" in the secondary gateways. Using the "realm" parameter seemed to help me, but it took experimentation to find what worked.
-------------------------------------------------------------------- <code>
<include> <gateway name="WhistlePhone"> <param name="username" value="xxxxxxxxxxx"/> <param name="from-user" value="xxxxxxxxxx"/> <param name="proxy" value="proxy.WhistlePhone.com"/> <param name="realm" value="WhistlePhone"/> <param name="password" value="xxxx"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway>
<gateway name="WhistlePhoneProxy1"> <param name="username" value="xxxxxxxxxxx"/> <param name="from-user" value="xxxxxxxxxx"/> <param name="proxy" value="proxy1.WhistlePhone.com"/> <param name="from-domain" value="proxy.Whistlephone.com"/> <param name="password" value="xxxx"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway>
repeat for each proxy server.....
</include>
</code> ------------------------------------------------------------------------
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Hi Patrick I've tried this, but FreeSWITCH still switch to the other server sometimes. It seams it use the url from the sip header. Any idea on how to solve this? Or is it a Bug? regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-304-... Sent from the freeswitch-users mailing list archive at Nabble.com.
On 11-06-12 17:16, sunsus wrote:
Hi Patrick
I've tried this, but FreeSWITCH still switch to the other server sometimes. It seams it use the url from the sip header.
Any idea on how to solve this? Or is it a Bug?
I'm afraid I don't know. If you can't figure it out perhaps some professional help from consulting@freeswitch.org could offer a solution. Regards, Patrick
working sip trace with debug: -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-304-... Sent from the freeswitch-users mailing list archive at Nabble.com.
No link, try again. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jun 12, 2012 2:08 PM, "sunsus" <patrick@sunsus.net> wrote:
working sip trace with debug:
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-304-... Sent from the freeswitch-users mailing list archive at Nabble.com.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Non working sip trace http://pastebin.freeswitch.org/19271 http://pastebin.freeswitch.org/19271 XML Config http://pastebin.freeswitch.org/19272 http://pastebin.freeswitch.org/19272 Working sip trace http://pastebin.freeswitch.org/19273 http://pastebin.freeswitch.org/19273 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-401-... Sent from the freeswitch-users mailing list archive at Nabble.com.
On 12/06/12 19:40, sunsus wrote:
Non working sip trace http://pastebin.freeswitch.org/19271 http://pastebin.freeswitch.org/19271 XML Config http://pastebin.freeswitch.org/19272 http://pastebin.freeswitch.org/19272 Working sip trace http://pastebin.freeswitch.org/19273 http://pastebin.freeswitch.org/19273
Try sending the call to 62.65.137.114 or to 62.65.137.113 instead of to sip.netvoip.ch and see if that works - it is does then the "issue" could be resolved by changes to the suppliers load balancer andor your DNS setup/cache. Changing the XML file above is not going to help you at the moment - it looks like your XML dialplan is still sending the call to sip.netvoip.ch, not to the IP address: Dialplan: sofia/internal/1003@pbx.domain.tld Action bridge([leg_progress_timeout=15]sofia/gateway/netvoip-41440000026/0041440000069@sip.netvoip.ch) I'd possibly suggest that your bridge command above looks wrong as well - you are using '/gateway' AND '@sip.netvoip.ch'... Regards,
Hello Paul Thanks for you answer, but now freeswitch is canceling the call: http://pastebin.freeswitch.org/19277 http://pastebin.freeswitch.org/19277 regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-401-... Sent from the freeswitch-users mailing list archive at Nabble.com.
On 12/06/12 20:58, sunsus wrote:
Hello Paul
Thanks for you answer, but now freeswitch is canceling the call: http://pastebin.freeswitch.org/19277 http://pastebin.freeswitch.org/19277
Looks like 1003@pbx is hanging up - can you adjust your logging to include the SIP signalling between (I presume) the handset and FreeSWITCH (the "internal" SIP profile)? We seem to be seeing FreeSWITCH<>Provider but not Handset<>FreeSWITCH? 2012-06-12 21:50:47.393989 [NOTICE] sofia.c:6618 Hangup sofia/internal/1003@pbx.domain.tld [CS_EXECUTE] [ORIGINATOR_CANCEL] Regards,
Hello Paul The Problem with the Hangup, was a policy issue on the phone because it rejected non tls calls. Sorry was my fault. But no I changes page everything to sip.netvoip.ch and changed the bridge to the gateway as you told me. But I have still the issue with authorization: Log: http://pastebin.freeswitch.org/19292 Gateway: http://pastebin.freeswitch.org/19294 bridge: <action application="bridge" data="sofia/gateway/netvoip-41440000000/0041$2$3" /> Do you really sure this is not a bug? regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-401-... Sent from the freeswitch-users mailing list archive at Nabble.com.
Hello Paul And if I change to the first server sip-1.netvoip.ch: freeswitch@internal> sofia_dig sip.netvoip.ch Preference Weight Transport Port Address ================================================================================ 1 0.100 udp 5060 62.65.137.113 1 0.100 udp 5060 62.65.137.114 everything works, so it seams to be a bug with a DNS Loadbalancing!? http://pastebin.freeswitch.org/19295 regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-401-... Sent from the freeswitch-users mailing list archive at Nabble.com.
Read this old thread. http://freeswitch-users.2379917.n2.nabble.com/Authorizations-when-using-DNS-... This is a configuration problem with dns on their side. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-401-... Sent from the freeswitch-users mailing list archive at Nabble.com.
Hello Jeff Thanks for the information, thats what I searched for. Now I know it's not a bug, but a provider problem. regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-401-... Sent from the freeswitch-users mailing list archive at Nabble.com.
not working sip trace with debug: -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-304-... Sent from the freeswitch-users mailing list archive at Nabble.com.
gateway configuration -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-304-... Sent from the freeswitch-users mailing list archive at Nabble.com.
participants (7)
-
Brian Foster
-
Jeff Lenk
-
Patrick Dürsteler
-
Patrick Lists
-
Paul Cupis
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sunsus
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Tom C