FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls. I tried configuring les.net (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG. I got gizmo to where status is REGED but from XLite when I dial a 10 digit number the console shows [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting Any tips ...or is there an affordable contractor out there who can get a "hello world" call going? sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= internal profile sip:mod_sofia@67.228.218.111:5060 RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com<sip%3A17476481805@proxy01.sipphone.com> REGED lesnet gateway sip:16643@did.voip.les.net<sip%3A16643@did.voip.les.net> NOREG 67.228.218.111 alias internal ALIASED outbound alias external ALIASED ================================================================================================= Thanks for any tips or suggestions! -Brian B
Can you send your dialplan and directory files. It sounds like you are calling into a context with no phones in it. Brian B wrote:
FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls.
I tried configuring les.net <http://les.net> (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG.
I got gizmo to where status is REGED but from XLite when I dial a 10 digit number the console shows [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting
Any tips ...or is there an affordable contractor out there who can get a "hello world" call going?
sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= internal profile sip:mod_sofia@67.228.218.111:5060 <http://sip:mod_sofia@67.228.218.111:5060> RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 <http://sip:mod_sofia@67.228.218.111:5070> RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 <http://sip:mod_sofia@67.228.218.111:5080> RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com <mailto:sip%3A17476481805@proxy01.sipphone.com> REGED lesnet gateway sip:16643@did.voip.les.net <mailto:sip%3A16643@did.voip.les.net> NOREG 67.228.218.111 <http://67.228.218.111> alias internal ALIASED outbound alias external ALIASED =================================================================================================
Thanks for any tips or suggestions!
-Brian B
------------------------------------------------------------------------
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Brian, Just registering doesn't input the routing info into the dialplan. You'll need to add it to the dialplan in addition to registration. Can you clarify how you did your registration? Maybe post your XML without user/pass in them so we can point out why it failed? /b On Jun 4, 2008, at 2:37 PM, Brian B wrote:
FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls.
I tried configuring les.net (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG.
I got gizmo to where status is REGED but from XLite when I dial a 10 digit number the console shows [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting
Any tips ...or is there an affordable contractor out there who can get a "hello world" call going?
sofia status API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = = ====================================================================== internal profile sip:mod_sofia@67.228.218.111:5060 RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com REGED lesnet gateway sip:16643@did.voip.les.net NOREG 67.228.218.111 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================
Thanks for any tips or suggestions!
-Brian B _______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi Brian, I created a LES profile located at sip_profiles/external/les.xml with following content: <include> <gateway name="did.voip.les.net"> <param name="username" value="1490236124"/> <param name="realm" value="did.voip.les.net"/> <param name="password" value="mash.n2rown4"/> </gateway> </include> My dialing is actually done in a Javascript file in the following way: session.execute("bridge","sofia/gateway/did.voip.les.net/14156113200"); Since you already have LES working with Asterisk, I guess you know about the settings that you need on the LES side. But you (or somebody else) might still want to know what is works for me. In the les.net management console I createa a Peer/Trunk. To do this follow the following steps: 1) Login at les.net 2) On the left menu click on "Peers / Trunk" 3) then click on "create peer" 4) Next click on "Edit" to edit the settings for this trunk. 5) Specify the IP Address and the password of the trunk. Hope this solves your problem. Klaus. -------- Original-Nachricht --------
Datum: Wed, 4 Jun 2008 12:37:51 -0700 Von: "Brian B" <freeswitch@brian-burt.com> An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] FS installed, but no external calls
FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls.
I tried configuring les.net (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG.
I got gizmo to where status is REGED but from XLite when I dial a 10 digit number the console shows [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting
Any tips ...or is there an affordable contractor out there who can get a "hello world" call going?
sofia status API CALL [sofia(status)] output: Name Type Data State ================================================================================================= internal profile sip:mod_sofia@67.228.218.111:5060 RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com<sip%3A17476481805@proxy01.sipphone.com> REGED lesnet gateway sip:16643@did.voip.les.net<sip%3A16643@did.voip.les.net> NOREG 67.228.218.111 alias internal ALIASED outbound alias external ALIASED =================================================================================================
Thanks for any tips or suggestions!
-Brian B
-- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger
how about you reply with the settings you currently use for Asterisk? /b On Jun 4, 2008, at 5:30 PM, Klaus Teller wrote:
Hi Brian,
I created a LES profile located at sip_profiles/external/les.xml with following content:
<include> <gateway name="did.voip.les.net"> <param name="username" value="1490236124"/> <param name="realm" value="did.voip.les.net"/> <param name="password" value="mash.n2rown4"/> </gateway> </include>
My dialing is actually done in a Javascript file in the following way:
session.execute("bridge","sofia/gateway/did.voip.les.net/ 14156113200");
Since you already have LES working with Asterisk, I guess you know about the settings that you need on the LES side. But you (or somebody else) might still want to know what is works for me.
In the les.net management console I createa a Peer/Trunk. To do this follow the following steps:
1) Login at les.net 2) On the left menu click on "Peers / Trunk" 3) then click on "create peer" 4) Next click on "Edit" to edit the settings for this trunk. 5) Specify the IP Address and the password of the trunk.
Hope this solves your problem.
Klaus.
-------- Original-Nachricht --------
Datum: Wed, 4 Jun 2008 12:37:51 -0700 Von: "Brian B" <freeswitch@brian-burt.com> An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] FS installed, but no external calls
FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls.
I tried configuring les.net (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG.
I got gizmo to where status is REGED but from XLite when I dial a 10 digit number the console shows [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting
Any tips ...or is there an affordable contractor out there who can get a "hello world" call going?
sofia status API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = = = ===================================================================== internal profile sip:mod_sofia@67.228.218.111:5060 RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com<sip%3A17476481805@proxy01.sipphone.com
REGED lesnet gateway sip:16643@did.voip.les.net<sip%3A16643@did.voip.les.net> NOREG 67.228.218.111 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = = = =====================================================================
Thanks for any tips or suggestions!
-Brian B
-- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Here are the Elastix/Asterisk settings that were working. Brian West remarked that I have to "add it to the dialplan in addition to registration". Can anyone give me a pointer on that? *Inbound* registration string: 16643:password@did.voip.les.net/16643 User context: from-trunk User Details: canreinvite=no context=from-trunk insecure=very nat=yes type=user *Outbound *canreinvite=no context=from-trunk dtmf=inband dtmfmode=inband fromuser=16643 host=did.voip.les.net insecure=very nat=yes qualify=yes secret=pass type=peer user=phone username=16643* * On Wed, Jun 4, 2008 at 4:08 PM, Brian West <brian@freeswitch.org> wrote:
how about you reply with the settings you currently use for Asterisk?
/b
On Jun 4, 2008, at 5:30 PM, Klaus Teller wrote:
Hi Brian,
I created a LES profile located at sip_profiles/external/les.xml with following content:
<include> <gateway name="did.voip.les.net"> <param name="username" value="1490236124"/> <param name="realm" value="did.voip.les.net"/> <param name="password" value="mash.n2rown4"/> </gateway> </include>
My dialing is actually done in a Javascript file in the following way:
session.execute("bridge","sofia/gateway/did.voip.les.net/ 14156113200");
Since you already have LES working with Asterisk, I guess you know about the settings that you need on the LES side. But you (or somebody else) might still want to know what is works for me.
In the les.net management console I createa a Peer/Trunk. To do this follow the following steps:
1) Login at les.net 2) On the left menu click on "Peers / Trunk" 3) then click on "create peer" 4) Next click on "Edit" to edit the settings for this trunk. 5) Specify the IP Address and the password of the trunk.
Hope this solves your problem.
Klaus.
-------- Original-Nachricht --------
Datum: Wed, 4 Jun 2008 12:37:51 -0700 Von: "Brian B" <freeswitch@brian-burt.com> An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] FS installed, but no external calls
FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls.
I tried configuring les.net (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG.
I got gizmo to where status is REGED but from XLite when I dial a 10 digit number the console shows [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting
Any tips ...or is there an affordable contractor out there who can get a "hello world" call going?
sofia status API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = = = ===================================================================== internal profile sip:mod_sofia@67.228.218.111:5060 RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com<sip%3A17476481805@proxy01.sipphone.com> <sip%3A17476481805@proxy01.sipphone.com<sip%253A17476481805@proxy01.sipphone.com>
REGED lesnet gateway sip:16643@did.voip.les.net <sip%3A16643@did.voip.les.net>< sip%3A16643@did.voip.les.net <sip%253A16643@did.voip.les.net>> NOREG 67.228.218.111 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = = = =====================================================================
Thanks for any tips or suggestions!
-Brian B
-- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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You'll need to set from-user on freeswitch also. /b On Jun 4, 2008, at 7:26 PM, Brian B wrote:
Here are the Elastix/Asterisk settings that were working. Brian West remarked that I have to "add it to the dialplan in addition to registration". Can anyone give me a pointer on that?
Inbound registration string: 16643:password@did.voip.les.net/16643 User context: from-trunk User Details: canreinvite=no context=from-trunk insecure=very nat=yes type=user
Outbound canreinvite=no context=from-trunk dtmf=inband dtmfmode=inband fromuser=16643 host=did.voip.les.net insecure=very nat=yes qualify=yes secret=pass type=peer user=phone username=16643
On Wed, Jun 4, 2008 at 4:08 PM, Brian West <brian@freeswitch.org> wrote: how about you reply with the settings you currently use for Asterisk?
/b
On Jun 4, 2008, at 5:30 PM, Klaus Teller wrote:
Hi Brian,
I created a LES profile located at sip_profiles/external/les.xml with following content:
<include> <gateway name="did.voip.les.net"> <param name="username" value="1490236124"/> <param name="realm" value="did.voip.les.net"/> <param name="password" value="mash.n2rown4"/> </gateway> </include>
My dialing is actually done in a Javascript file in the following way:
session.execute("bridge","sofia/gateway/did.voip.les.net/ 14156113200");
Since you already have LES working with Asterisk, I guess you know about the settings that you need on the LES side. But you (or somebody else) might still want to know what is works for me.
In the les.net management console I createa a Peer/Trunk. To do this follow the following steps:
1) Login at les.net 2) On the left menu click on "Peers / Trunk" 3) then click on "create peer" 4) Next click on "Edit" to edit the settings for this trunk. 5) Specify the IP Address and the password of the trunk.
Hope this solves your problem.
Klaus.
-------- Original-Nachricht --------
Datum: Wed, 4 Jun 2008 12:37:51 -0700 Von: "Brian B" <freeswitch@brian-burt.com> An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] FS installed, but no external calls
FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls.
I tried configuring les.net (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG.
I got gizmo to where status is REGED but from XLite when I dial a 10 digit number the console shows [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting
Any tips ...or is there an affordable contractor out there who can get a "hello world" call going?
sofia status API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = = =
=====================================================================
internal profile sip:mod_sofia@67.228.218.111:5060 RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com<sip%3A17476481805@proxy01.sipphone.com
REGED lesnet gateway sip:16643@did.voip.les.net<sip%3A16643@did.voip.les.net> NOREG 67.228.218.111 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = = =
=====================================================================
Thanks for any tips or suggestions!
-Brian B
-- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger
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Now the console is showing that there's an inbound call, but I don't know how to route it to an extension, which I'm gathering is in the dialplan? There were some "advanced" examples in the documentation but nothing with something simple like "route all calls to extension 1001" - which I will plan to add to the wiki when I get this figured out! I admit to being a complete nube - barely know linux, regex etc. My contribution will be in documentation for nubes like myself! :-) Here's the console log: 2008-06-05 01:07:04 [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel sofia/external/4153084258@64.34.181.47[38a9fd64-dd73-4b6a-9cd9-9cd4675261ea] # that is my inbound DID 2008-06-05 01:07:04 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing NOT FOUND->4153581872@public 2008-06-05 01:07:04 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-06-05 01:07:04 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/external/ 4153084258@64.34.181.47 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-06-05 01:07:04 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 8 (sofia/external/ 4153084258@64.34.181.47) Ended 2008-06-05 01:07:04 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/external/ 4153084258@64.34.181.47 [CS_HANGUP] On Wed, Jun 4, 2008 at 5:36 PM, Brian West <brian@freeswitch.org> wrote:
You'll need to set from-user on freeswitch also. /b
On Jun 4, 2008, at 7:26 PM, Brian B wrote:
Here are the Elastix/Asterisk settings that were working. Brian West remarked that I have to "add it to the dialplan in addition to registration". Can anyone give me a pointer on that?
*Inbound* registration string: 16643:password@did.voip.les.net/16643 User context: from-trunk User Details: canreinvite=no context=from-trunk insecure=very nat=yes type=user
*Outbound *canreinvite=no context=from-trunk dtmf=inband dtmfmode=inband fromuser=16643 host=did.voip.les.net insecure=very nat=yes qualify=yes secret=pass type=peer user=phone username=16643* * On Wed, Jun 4, 2008 at 4:08 PM, Brian West <brian@freeswitch.org> wrote:
how about you reply with the settings you currently use for Asterisk?
/b
On Jun 4, 2008, at 5:30 PM, Klaus Teller wrote:
Hi Brian,
I created a LES profile located at sip_profiles/external/les.xml with following content:
<include> <gateway name="did.voip.les.net"> <param name="username" value="1490236124"/> <param name="realm" value="did.voip.les.net"/> <param name="password" value="mash.n2rown4"/> </gateway> </include>
My dialing is actually done in a Javascript file in the following way:
session.execute("bridge","sofia/gateway/did.voip.les.net/ 14156113200");
Since you already have LES working with Asterisk, I guess you know about the settings that you need on the LES side. But you (or somebody else) might still want to know what is works for me.
In the les.net management console I createa a Peer/Trunk. To do this follow the following steps:
1) Login at les.net 2) On the left menu click on "Peers / Trunk" 3) then click on "create peer" 4) Next click on "Edit" to edit the settings for this trunk. 5) Specify the IP Address and the password of the trunk.
Hope this solves your problem.
Klaus.
-------- Original-Nachricht --------
Datum: Wed, 4 Jun 2008 12:37:51 -0700 Von: "Brian B" <freeswitch@brian-burt.com> An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] FS installed, but no external calls
FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls.
I tried configuring les.net (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG.
I got gizmo to where status is REGED but from XLite when I dial a 10 digit number the console shows [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting
Any tips ...or is there an affordable contractor out there who can get a "hello world" call going?
sofia status API CALL [sofia(status)] output: Name Type Data State = = = = = = = = = = = = = = = = = = = = = = = = = = = = ===================================================================== internal profile sip:mod_sofia@67.228.218.111:5060 RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com<sip%3A17476481805@proxy01.sipphone.com> <sip%3A17476481805@proxy01.sipphone.com<sip%253A17476481805@proxy01.sipphone.com>
REGED lesnet gateway sip:16643@did.voip.les.net <sip%3A16643@did.voip.les.net>< sip%3A16643@did.voip.les.net <sip%253A16643@did.voip.les.net>> NOREG 67.228.218.111 alias internal ALIASED outbound alias external ALIASED = = = = = = = = = = = = = = = = = = = = = = = = = = = = =====================================================================
Thanks for any tips or suggestions!
-Brian B
-- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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In public.xml, you want something like this: <extension name="test_did"> <condition field="destination_number" expression="^(4153084258)$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> and then in default.xml have something like this in the default context: <extension name="Local_Extension"> <condition field="destination_number" expression="^(4153084258)$" continue= "on-true"> <action application="set" data="dialed_ext=$1"/> </condition> <condition field="destination_number" expression="^${caller_id_number}$"> <action application="set" data="voicemail_authorized=${sip_authorized}"/
<action application="answer"/> <action application="sleep" data="1000"/> <action application="voicemail" data="check default $${domain} ${dialed_ ext}"/> <anti-action application="ring_ready"/> <anti-action application="set" data="call_timeout=10"/> <anti-action application="set" data="hangup_after_bridge=true"/> <anti-action application="set" data="continue_on_fail=true"/> <anti-action application="bridge" data="USER/${dialed_ext}@$${domain}"/> <anti-action application="answer"/> <anti-action application="sleep" data="1000"/> <anti-action application="voicemail" data="default $${domain} ${dialed_e xt}"/> </condition> </extension> Brian B wrote:
Now the console is showing that there's an inbound call, but I don't know how to route it to an extension, which I'm gathering is in the dialplan? There were some "advanced" examples in the documentation but nothing with something simple like "route all calls to extension 1001" - which I will plan to add to the wiki when I get this figured out!
I admit to being a complete nube - barely know linux, regex etc. My contribution will be in documentation for nubes like myself! :-)
Here's the console log:
2008-06-05 01:07:04 [NOTICE] switch_channel.c:533 switch_channel_set_name() New Channel sofia/external/4153084258@64.34.181.47 <mailto:4153084258@64.34.181.47> [38a9fd64-dd73-4b6a-9cd9-9cd4675261ea] # that is my inbound DID 2008-06-05 01:07:04 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing NOT FOUND->4153581872@public 2008-06-05 01:07:04 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-06-05 01:07:04 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/external/4153084258@64.34.181.47 <mailto:4153084258@64.34.181.47> [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-06-05 01:07:04 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 8 (sofia/external/4153084258@64.34.181.47 <mailto:4153084258@64.34.181.47>) Ended 2008-06-05 01:07:04 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/external/4153084258@64.34.181.47 <mailto:4153084258@64.34.181.47> [CS_HANGUP]
On Wed, Jun 4, 2008 at 5:36 PM, Brian West <brian@freeswitch.org <mailto:brian@freeswitch.org>> wrote:
You'll need to set from-user on freeswitch also.
/b
On Jun 4, 2008, at 7:26 PM, Brian B wrote:
Here are the Elastix/Asterisk settings that were working. Brian West remarked that I have to "add it to the dialplan in addition to registration". Can anyone give me a pointer on that?
*Inbound* registration string: 16643:password@did.voip.les.net/16643 <http://16643:password@did.voip.les.net/16643> User context: from-trunk User Details: canreinvite=no context=from-trunk insecure=very nat=yes type=user
*Outbound *canreinvite=no context=from-trunk dtmf=inband dtmfmode=inband fromuser=16643 host=did.voip.les.net <http://did.voip.les.net> insecure=very nat=yes qualify=yes secret=pass type=peer user=phone username=16643* * On Wed, Jun 4, 2008 at 4:08 PM, Brian West <brian@freeswitch.org <mailto:brian@freeswitch.org>> wrote:
how about you reply with the settings you currently use for Asterisk?
/b
On Jun 4, 2008, at 5:30 PM, Klaus Teller wrote:
> Hi Brian, > > I created a LES profile located at sip_profiles/external/les.xml > with following content: > > <include> > <gateway name="did.voip.les.net <http://did.voip.les.net>"> > <param name="username" value="1490236124"/> > <param name="realm" value="did.voip.les.net <http://did.voip.les.net>"/> > <param name="password" value="mash.n2rown4"/> > </gateway> > </include> > > > My dialing is actually done in a Javascript file in the following way: > > session.execute("bridge","sofia/gateway/did.voip.les.net/ <http://did.voip.les.net/> > 14156113200"); > > Since you already have LES working with Asterisk, I guess you know > about the settings that you need on the LES side. But you (or > somebody else) might still want to know what is works for me. > > > In the les.net <http://les.net> management console I createa a Peer/Trunk. To do > this follow the following steps: > > 1) Login at les.net <http://les.net> > 2) On the left menu click on "Peers / Trunk" > 3) then click on "create peer" > 4) Next click on "Edit" to edit the settings for this trunk. > 5) Specify the IP Address and the password of the trunk. > > > Hope this solves your problem. > > Klaus. > > > > > > > > > > > > > -------- Original-Nachricht -------- >> Datum: Wed, 4 Jun 2008 12:37:51 -0700 >> Von: "Brian B" <freeswitch@brian-burt.com <mailto:freeswitch@brian-burt.com>> >> An: freeswitch-users@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org> >> Betreff: [Freeswitch-users] FS installed, but no external calls > >> FS is definitely up and running on my Lylix VPS, which is exciting. >> X-Lite >> is dialed in and can talk to the IVR etc. But no inbound or outbound >> calls. >> >> I tried configuring les.net <http://les.net> (which I got working in * on the same >> box) but >> couldn't get that working, and sofia status is NOREG. >> >> I got gizmo to where status is REGED but from XLite when I dial a >> 10 digit >> number the console shows >> [INFO] switch_core_state_machine.c:114 >> switch_core_standard_on_routing() >> No >> Route, Aborting >> >> >> Any tips ...or is there an affordable contractor out there who can >> get a >> "hello world" call going? >> >> >> >> sofia status >> API CALL [sofia(status)] output: >> Name Type >> Data State >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> ===================================================================== >> internal profile >> sip:mod_sofia@67.228.218.111:5060 <http://sip:mod_sofia@67.228.218.111:5060> >> RUNNING (0) >> nat profile >> sip:mod_sofia@67.228.218.111:5070 <http://sip:mod_sofia@67.228.218.111:5070> >> RUNNING (0) >> default alias >> internal ALIASED >> external profile >> sip:mod_sofia@67.228.218.111:5080 <http://sip:mod_sofia@67.228.218.111:5080> >> RUNNING (0) >> gizmo gateway >> sip:17476481805@proxy01.sipphone.com <mailto:sip%3A17476481805@proxy01.sipphone.com><sip%3A17476481805@proxy01.sipphone.com <mailto:sip%253A17476481805@proxy01.sipphone.com> >> > >> REGED >> lesnet gateway >> sip:16643@did.voip.les.net <mailto:sip%3A16643@did.voip.les.net><sip%3A16643@did.voip.les.net <mailto:sip%253A16643@did.voip.les.net>> >> NOREG >> 67.228.218.111 <http://67.228.218.111> alias >> internal ALIASED >> outbound alias >> external ALIASED >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> = >> ===================================================================== >> >> >> Thanks for any tips or suggestions! >> >> -Brian B > > -- > Psssst! Schon vom neuen GMX MultiMessenger gehört? > Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users@lists.freeswitch.org <mailto:Freeswitch-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org
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------------------------------------------------------------------------
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FYI, calls from the "public" go into the public context where they then need to be transferred to another more friendly context for processing, like default. That is why you add the entry to public and the 'data="$1 XML PFC"' says to transfer called number $1 to the context PFC using XML dialplan. In the default context is where you actually ring the phone. Peder
Here's my setup: conf/sip_profiles/outbound/lesnet.xml <include> <gateway name="lesnet"> <param name="username" value="1659525395"/> <param name="password" value="XXXXXX"/> <param name="realm" value="did.voip.les.net"/> <param name="proxy" value="did.voip.les.net"/> <param name="register" value="true"/> </gateway> </include> Then: sessionx.originate(session, "sofia/gateway/lesnet/1502XXXXXXX", 30); HTH, Nick On Wed, Jun 4, 2008 at 6:30 PM, Klaus Teller <klaus.teller@gmx.net> wrote:
Hi Brian,
I created a LES profile located at sip_profiles/external/les.xml with following content:
<include> <gateway name="did.voip.les.net"> <param name="username" value="1490236124"/> <param name="realm" value="did.voip.les.net"/> <param name="password" value="mash.n2rown4"/> </gateway> </include>
My dialing is actually done in a Javascript file in the following way:
session.execute("bridge","sofia/gateway/did.voip.les.net/14156113200");
Since you already have LES working with Asterisk, I guess you know about the settings that you need on the LES side. But you (or somebody else) might still want to know what is works for me.
In the les.net management console I createa a Peer/Trunk. To do this follow the following steps:
1) Login at les.net 2) On the left menu click on "Peers / Trunk" 3) then click on "create peer" 4) Next click on "Edit" to edit the settings for this trunk. 5) Specify the IP Address and the password of the trunk.
Hope this solves your problem.
Klaus.
-------- Original-Nachricht --------
Datum: Wed, 4 Jun 2008 12:37:51 -0700 Von: "Brian B" <freeswitch@brian-burt.com> An: freeswitch-users@lists.freeswitch.org Betreff: [Freeswitch-users] FS installed, but no external calls
FS is definitely up and running on my Lylix VPS, which is exciting. X-Lite is dialed in and can talk to the IVR etc. But no inbound or outbound calls.
I tried configuring les.net (which I got working in * on the same box) but couldn't get that working, and sofia status is NOREG.
I got gizmo to where status is REGED but from XLite when I dial a 10 digit number the console shows [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting
Any tips ...or is there an affordable contractor out there who can get a "hello world" call going?
sofia status API CALL [sofia(status)] output: Name Type Data State
=================================================================================================
internal profile sip:mod_sofia@67.228.218.111:5060 RUNNING (0) nat profile sip:mod_sofia@67.228.218.111:5070 RUNNING (0) default alias internal ALIASED external profile sip:mod_sofia@67.228.218.111:5080 RUNNING (0) gizmo gateway sip:17476481805@proxy01.sipphone.com<sip%3A17476481805@proxy01.sipphone.com>
<sip%3A17476481805@proxy01.sipphone.com<sip%253A17476481805@proxy01.sipphone.com>
REGED lesnet gateway sip:16643@did.voip.les.net <sip%3A16643@did.voip.les.net><
sip%3A16643@did.voip.les.net <sip%253A16643@did.voip.les.net>>
NOREG 67.228.218.111 alias internal ALIASED outbound alias external ALIASED
=================================================================================================
Thanks for any tips or suggestions!
-Brian B
-- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger
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I just want to point out one thing.. if you called the gateway did.voip.les.net then the ream and proxy values can be left out. We will just use that. /b On Jun 4, 2008, at 6:14 PM, Nick Temple wrote:
<include> <gateway name="lesnet"> <param name="username" value="1659525395"/> <param name="password" value="XXXXXX"/> <param name="realm" value="did.voip.les.net"/> <param name="proxy" value="did.voip.les.net"/> <param name="register" value="true"/> </gateway> </include>
thanks for the pointer ... so the only difference between the below and Brian B's is the register param? On Wed, Jun 4, 2008 at 7:20 PM, Brian West <brian@freeswitch.org> wrote:
I just want to point out one thing.. if you called the gateway did.voip.les.net then the ream and proxy values can be left out. We will just use that. /b
On Jun 4, 2008, at 6:14 PM, Nick Temple wrote:
<include> <gateway name="lesnet"> <param name="username" value="1659525395"/> <param name="password" value="XXXXXX"/> <param name="realm" value="did.voip.les.net"/> <param name="proxy" value="did.voip.les.net"/> <param name="register" value="true"/> </gateway> </include>
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participants (5)
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Brian B
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Brian West
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Klaus Teller
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Nick Temple
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Peder ＠ NetworkOblivion