Hi Matt,
I redirected this email on the users mailing list - it's more appropriate.
the idea seams ok, with couple of comments: 1) be sure that fwd to localhost is ok (instead of a routable IP) 2) doing Record-Route may be a good think.
to debug tour problem, add some log("...") statements into your script to be able to trace the processing. Also a network trace (including on lo device) will be helpful to see what happens - if the messages are received, if they are sent and where. Also watch the log for potential errors.
regards, bogdan
Matt L. Zhu wrote:
has anyone successfully setup openser as the frontend proxy for asterisk? here is my setup
/etc/asterisk/sip.conf [general] context=default port=5065 bindaddr=0.0.0.0 srvlookup=yes
[ser] type=user context=proxy host=192.168.0.10
then i edited openser.cfg to do something like this
if
(uri=~"sip:[a-zA-Z.]*@(xxx.xxx.com)|(192.168.0.10)") { forward( localhost, 5065 ); break; };
i connected two sipphones (wengo) in this case to openser, but calls are not going through at all, connecting directly to asterisk works. have anyone worked in this situation?
thanks
Devel mailing list Devel@openser.org http://openser.org/cgi-bin/mailman/listinfo/devel
whats is sip debug on asterisk showing
Bogdan-Andrei Iancu wrote:
Hi Matt,
I redirected this email on the users mailing list - it's more appropriate.
the idea seams ok, with couple of comments:
- be sure that fwd to localhost is ok (instead of a routable IP)
- doing Record-Route may be a good think.
to debug tour problem, add some log("...") statements into your script to be able to trace the processing. Also a network trace (including on lo device) will be helpful to see what happens - if the messages are received, if they are sent and where. Also watch the log for potential errors.
regards, bogdan
Matt L. Zhu wrote:
has anyone successfully setup openser as the frontend proxy for asterisk? here is my setup
/etc/asterisk/sip.conf [general] context=default port=5065 bindaddr=0.0.0.0 srvlookup=yes
[ser] type=user context=proxy host=192.168.0.10
then i edited openser.cfg to do something like this
if
(uri=~"sip:[a-zA-Z.]*@(xxx.xxx.com)|(192.168.0.10)") { forward( localhost, 5065 ); break; };
i connected two sipphones (wengo) in this case to openser, but calls are not going through at all, connecting directly to asterisk works. have anyone worked in this situation?
thanks
Devel mailing list Devel@openser.org http://openser.org/cgi-bin/mailman/listinfo/devel
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
.
You may want to set type=peer in the [ser] section. Also , I assume you have a Dial statement in your 'proxy' context in the dialplan. You need that to connect the 2 users. We have no problems using Asterisk as a sip server with ser or openser as the registrar and proxy. I think there are many using this kind of setup so it does work.
Mark
On 9/29/05, Iqbal iqbal@gigo.co.uk wrote:
whats is sip debug on asterisk showing
Bogdan-Andrei Iancu wrote:
Hi Matt,
I redirected this email on the users mailing list - it's more appropriate.
the idea seams ok, with couple of comments:
- be sure that fwd to localhost is ok (instead of a routable IP)
- doing Record-Route may be a good think.
to debug tour problem, add some log("...") statements into your script to be able to trace the processing. Also a network trace (including on lo device) will be helpful to see what happens - if the messages are received, if they are sent and where. Also watch the log for potential errors.
regards, bogdan
Matt L. Zhu wrote:
has anyone successfully setup openser as the frontend proxy for asterisk? here is my setup
/etc/asterisk/sip.conf [general] context=default port=5065 bindaddr=0.0.0.0 http://0.0.0.0 srvlookup=yes
[ser] type=user context=proxy host=192.168.0.10 http://192.168.0.10
then i edited openser.cfg to do something like this
if (uri=~"sip:[a-zA-Z.]*@(xxx.xxx.com)|(192.168.0.10)") { forward( localhost, 5065 ); break; };
i connected two sipphones (wengo) in this case to openser, but calls are not going through at all, connecting directly to asterisk works. have anyone worked in this situation?
thanks
Devel mailing list Devel@openser.org http://openser.org/cgi-bin/mailman/listinfo/devel
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
.
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
in my dialplan, i have this
[proxy] # same as the context in sip.conf exten => 4005.,1,Dial(SIP/${EXTEN}@192.168.0.10)
i am new to asterisk, how can i make it so the exten will route the call to the other sipphone connected to the ser proxy.
i really want to achieve sipphone->ser->asterisk->ser->sipphone when a phone calls another. just getting confused how exten will reroute to ser again.
<BLOCKQUOTE style='PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #A0C6E5 2px solid; MARGIN-RIGHT: 0px'><font style='FONT-SIZE:11px;FONT-FAMILY:tahoma,sans-serif'><hr color=#A0C6E5 size=1> From: <i>Mark Aiken <aiken.mark@gmail.com></i><br>Reply-To: <i>Mark Aiken <aiken.mark@gmail.com></i><br>To: <i>Iqbal <iqbal@gigo.co.uk></i><br>CC: <i>Bogdan-Andrei Iancu <bogdan@voice-system.ro>, "Matt L. Zhu" <coder0000@hotmail.com>, users@openser.org</i><br>Subject: <i>Re: [Users] Re: [Devel] openser and asterisk</i><br>Date: <i>Thu, 29 Sep 2005 12:04:21 -0500</i><br> <br>You may want to set type=peer in the [ser] section. Also , I assume you have a Dial statement in your 'proxy' context in the dialplan. You need that to connect the 2 users. We have no problems using Asterisk as a sip server with ser or openser as the registrar and proxy. I think there are many using this kind of setup so it does work.<br> <br> Mark<br><br><div><span class="gmail_quote">On 9/29/05, <b class="gmail_sendername">Iqbal</b> <<a href="mailto:iqbal@gigo.co.uk">iqbal@gigo.co.uk</a>> wrote:</span><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;padding-left:1ex"> whats is sip debug on asterisk showing<br><br>Bogdan-Andrei Iancu wrote:<br><br>> Hi Matt,<br>><br>> I redirected this email on the users mailing list - it's more<br>> appropriate.<br>><br>> the idea seams ok, with couple of comments: <br>> 1) be sure that fwd to localhost is ok (instead of a routable IP)<br>> 2) doing Record-Route may be a good think.<br>><br>> to debug tour problem, add some log("...") statements into your script <br>> to be able to trace the processing. Also a network trace (including on<br>> lo device) will be helpful to see what happens - if the messages are<br>> received, if they are sent and where. Also watch the log for potential <br>> errors.<br>><br>> regards,<br>> bogdan<br>><br>><br>><br>> Matt L. Zhu wrote:<br>><br>>> has anyone successfully setup openser as the frontend proxy for<br>>> asterisk? here is my setup <br>>><br>>> /etc/asterisk/sip.conf<br>>> [general]<br>>> context=default<br>>> port=5065<br>>> bindaddr=<a href="http://0.0.0.0">0.0.0.0</a><br>>> srvlookup=yes<br>>><br>>> [ser] <br>>> type=user<br>>> context=proxy<br>>> host=<a href="http://192.168.0.10">192.168.0.10</a><br>>><br>>> then i edited openser.cfg to do something like this<br>>><br>>> if <br>>> (uri=~"sip:[a-zA-Z.]*@(xxx.xxx.com)|(192.168.0.10)") {<br>>> forward( localhost, 5065 );<br>>> break;<br>>> };<br>>><br>>> i connected two sipphones (wengo) in this case to openser, but calls<br>>> are not going through at all, connecting directly to asterisk works. <br>>> have anyone worked in this situation?<br>>><br>>> thanks<br>>><br>>><br>>><br>>> _______________________________________________<br>>> Devel mailing list<br>>> <a href="mailto:Devel@openser.org">Devel@openser.org</a><br>>> <a href="http://openser.org/cgi-bin/mailman/listinfo/devel">http://openser.org/cgi-bin/mailman/listinfo/devel</a><br>>><br>><br>><br>> _______________________________________________ <br>> Users mailing list<br>> <a href="mailto:Users@openser.org">Users@openser.org</a><br>> <a href="http://openser.org/cgi-bin/mailman/listinfo/users">http://openser.org/cgi-bin/mailman/listinfo/users</a><br>> <br>> .<br>><br><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@openser.org">Users@openser.org</a><br><a href="http://openser.org/cgi-bin/mailman/listinfo/users">http://openser.org/cgi-bin/mailman/listinfo/users </a><br></blockquote></div><br>
<br></font></BLOCKQUOTE>
Matt,
You must have user=peer (I think friend will also work) in the [ser] section of the sip.conf in order for Asterisk to find and dial that 4005 extension in your [proxy] context.
That exten should then setup a SIP call to 4005@192.168.0.10 for an incoming RURI of 4005@xxxxxxxxxx. Assuming that IP is SER then SER will need to do a lookup for that user and domain to find the contact address. The outgoing RURI to SER for the 2 leg will be 4005@192.168.0.10. Your SER setup need to deal with routing the calls via Asterisk in this way.
A general. complete solution with SER as registrar and proxy and Asterisk as B2BUA Feature Server for SIP PBX is beyond the scope of this email :). I've done this for a client under NDA so I cant just publish all of their ser and Asterisk configs. I dont own them. Perhaps there are other on this list that are free to publish that sort of very detailed configuration info.
I've found it very difficult to find specific help for difficult problems on mailing lists. Most of the hard questions go unanswered, likely due to NDAs and conflicts of interest. A consultant that puts food on the table by supporting open-source software is unlikely to give his 'product" away for free.
The SER community seems much better about this and the core developers answer a lot of hard questions on the various mailing lists.
The "core" developers for Asterisk are Digium and they sell support for a living so dont expect too much free support. Getting the software for free is a pretty good bargin , IMO, even if there is no 'free' support.
Mark
On 9/29/05, Matt L. Zhu coder0000@hotmail.com wrote:
in my dialplan, i have this
[proxy] # same as the context in sip.conf exten => 4005.,1,Dial(SIP/${EXTEN}@192.168.0.10)
i am new to asterisk, how can i make it so the exten will route the call to the other sipphone connected to the ser proxy.
i really want to achieve sipphone->ser->asterisk->ser->sipphone when a phone calls another. just getting confused how exten will reroute to ser again.
<BLOCKQUOTE style='PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #A0C6E5 2px solid; MARGIN-RIGHT: 0px'><font style='FONT-SIZE:11px;FONT-FAMILY:tahoma,sans-serif'><hr color=#A0C6E5 size=1> From: <i>Mark Aiken <aiken.mark@gmail.com></i><br>Reply-To: <i>Mark Aiken <aiken.mark@gmail.com></i><br>To: <i>Iqbal <iqbal@gigo.co.uk></i><br>CC: <i>Bogdan-Andrei Iancu <bogdan@voice-system.ro>, "Matt L. Zhu" <coder0000@hotmail.com>, users@openser.org</i><br>Subject: <i>Re: [Users] Re: [Devel] openser and asterisk</i><br>Date: <i>Thu, 29 Sep 2005 12:04:21 -0500</i><br> <br>You may want to set type=peer in the [ser] section. Also , I assume you have a Dial statement in your 'proxy' context in the dialplan. You need that to connect the 2 users. We have no problems using Asterisk as a sip server with ser or openser as the registrar and proxy. I think there are many using this kind of setup so it does work.<br> <br> Mark<br><br><div><span class="gmail_quote">On 9/29/05, <b class="gmail_sendername">Iqbal</b> <<a href="mailto:iqbal@gigo.co.uk">iqbal@gigo.co.uk</a>> wrote:</span><blockquote class="gmail_quote" style="margin:0pt 0pt 0pt 0.8ex;padding-left:1ex"> whats is sip debug on asterisk showing<br><br>Bogdan-Andrei Iancu wrote:<br><br>> Hi Matt,<br>><br>> I redirected this email on the users mailing list - it's more<br>> appropriate.<br>><br>> the idea seams ok, with couple of comments: <br>> 1) be sure that fwd to localhost is ok (instead of a routable IP)<br>> 2) doing Record-Route may be a good think.<br>><br>> to debug tour problem, add some log("...") statements into your script <br>> to be able to trace the processing. Also a network trace (including on<br>> lo device) will be helpful to see what happens - if the messages are<br>> received, if they are sent and where. Also watch the log for potential <br>> errors.<br>><br>> regards,<br>> bogdan<br>><br>><br>><br>> Matt L. Zhu wrote:<br>><br>>> has anyone successfully setup openser as the frontend proxy for<br>>> asterisk? here is my setup <br>>><br>>> /etc/asterisk/sip.conf<br>>> [general]<br>>> context=default<br>>> port=5065<br>>> bindaddr=<a href="http://0.0.0.0">0.0.0.0 <http://0.0.0.0></a><br>>> srvlookup=yes<br>>><br>>> [ser] <br>>> type=user<br>>> context=proxy<br>>> host=<a href="http://192.168.0.10">192.168.0.10 <http://192.168.0.10></a><br>>><br>>> then i edited openser.cfg to do something like this<br>>><br>>> if <br>>> (uri=~"sip:[a-zA-Z\.]*@(xxx\.xxx\.com)|(192\.168\.0\.10)") {<br>>> forward( localhost, 5065 );<br>>> break;<br>>> };<br>>><br>>> i connected two sipphones (wengo) in this case to openser, but calls<br>>> are not going through at all, connecting directly to asterisk works. <br>>> have anyone worked in this situation?<br>>><br>>> thanks<br>>><br>>><br>>><br>>> _______________________________________________<br>>> Devel mailing list<br>>> <a href="mailto:Devel@openser.org">Devel@openser.org</a><br>>> <a href="http://openser.org/cgi-bin/mailman/listinfo/devel"> http://openser.org/cgi-bin/mailman/listinfo/devel</a><br>>><br>><br>><br>> _______________________________________________ <br>> Users mailing list<br>> <a href="mailto:Users@openser.org">Users@openser.org</a><br>> <a href="http://openser.org/cgi-bin/mailman/listinfo/users"> http://openser.org/cgi-bin/mailman/listinfo/users</a><br>> <br>> .<br>><br><br>_______________________________________________<br>Users mailing list<br><a href="mailto:Users@openser.org">Users@openser.org</a><br><a href="http://openser.org/cgi-bin/mailman/listinfo/users"> http://openser.org/cgi-bin/mailman/listinfo/users </a><br></blockquote></div><br>
<br></font></BLOCKQUOTE>
hi,
has anyone successfully implemented the jabber module for openser? i keep on getting this error the moment i enable jabber.so in openser.cfg
1(0) get_connection(): Inherited open database connections, this is not a good idea 1(0) db_init(): Could not create a connection 1(0) ERROR: child_init(-2): Error while connecting database 1(0) init_mod_child(): Error while initializing module usrloc 1(0) ERROR: open_uac_fifo: init_child failed 1(0) starting fifo server failed 1(0) XJAB: Unloading module ...
without jabber, everything works fine, here are my parameters. modparam("jabber", "db_url", "mysql://root:root@localhost/sip") modparam("jabber", "jaddress", "192.168.0.4") modparam("jabber", "jdomain", "192.168.0.4") modparam("jabber", "proxy", "127.0.0.1:5060 localhost:5060")
please help
Hello,
On 10/05/05 00:52, Matt L. Zhu wrote:
hi,
has anyone successfully implemented the jabber module for openser? i keep on getting this error the moment i enable jabber.so in openser.cfg
1(0) get_connection(): Inherited open database connections, this is not a good idea 1(0) db_init(): Could not create a connection 1(0) ERROR: child_init(-2): Error while connecting database 1(0) init_mod_child(): Error while initializing module usrloc
the error comes from the usrloc module. OpenSER was unable to open the connection to MySQL server. Check the username, password, host and database you gave to db_url parameter of usrloc module.
Daniel
1(0) ERROR: open_uac_fifo: init_child failed 1(0) starting fifo server failed 1(0) XJAB: Unloading module ...
without jabber, everything works fine, here are my parameters. modparam("jabber", "db_url", "mysql://root:root@localhost/sip") modparam("jabber", "jaddress", "192.168.0.4") modparam("jabber", "jdomain", "192.168.0.4") modparam("jabber", "proxy", "127.0.0.1:5060 localhost:5060")
please help
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
hi daniel,
the problem doesn't seem to be usrloc, if i comment out the configuration associated with jabber, everything is fine, if i uncomment jabber, usrloc error happens.
<BLOCKQUOTE style='PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #A0C6E5 2px solid; MARGIN-RIGHT: 0px'><font style='FONT-SIZE:11px;FONT-FAMILY:tahoma,sans-serif'><hr color=#A0C6E5 size=1> From: <i>Daniel-Constantin Mierla <daniel@voice-system.ro></i><br>Reply-To: <i>daniel@voice-system.ro</i><br>To: <i>"Matt L. Zhu" <coder0000@hotmail.com></i><br>CC: <i>users@openser.org</i><br>Subject: <i>Re: [Users] openser and jabber</i><br>Date: <i>Wed, 05 Oct 2005 11:32:24 +0300</i><br>>Hello,<br>><br>>On 10/05/05 00:52, Matt L. Zhu wrote:<br>><br>>>hi,<br>>><br>>>has anyone successfully implemented the jabber module for openser? <br>>>i keep on getting this error the moment i enable jabber.so in <br>>>openser.cfg<br>>><br>>>1(0) get_connection(): Inherited open database connections, this is <br>>>not a good idea<br>>>1(0) db_init(): Could not create a connection<br>>>1(0) ERROR: child_init(-2): Error while connecting database<br>>>1(0) init_mod_child(): Error while initializing module usrloc<br>><br>>the error comes from the usrloc module. OpenSER was unable to open <br>>the connection to MySQL server. Check the username, password, host <br>>and database you gave to db_url parameter of usrloc module.<br>><br>>Daniel<br>><br>>>1(0) ERROR: open_uac_fifo: init_child failed<br>>>1(0) starting fifo server failed<br>>>1(0) XJAB: Unloading module ...<br>>><br>>><br>>>without jabber, everything works fine, here are my parameters.<br>>>modparam("jabber", "db_url", "mysql://root:root@localhost/sip")<br>>>modparam("jabber", "jaddress", "192.168.0.4")<br>>>modparam("jabber", "jdomain", "192.168.0.4")<br>>>modparam("jabber", "proxy", "127.0.0.1:5060 localhost:5060")<br>>><br>>>please help<br>>><br>>><br>>><br>>>_______________________________________________<br>>>Users mailing list<br>>>Users@openser.org<br>>>http://openser.org/cgi-bin/mailman/listinfo/users<br>>><br></font></BLOCKQUOTE>
i think the issue is usrloc and jabber are trying to share the same connection and creating an error. if i use dbmode=0 for usrloc, the error goes away, but i want to use persist database for usrloc as well.. is there a workaround?
<BLOCKQUOTE style='PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #A0C6E5 2px solid; MARGIN-RIGHT: 0px'><font style='FONT-SIZE:11px;FONT-FAMILY:tahoma,sans-serif'><hr color=#A0C6E5 size=1> From: <i>Daniel-Constantin Mierla <daniel@voice-system.ro></i><br>Reply-To: <i>daniel@voice-system.ro</i><br>To: <i>"Matt L. Zhu" <coder0000@hotmail.com></i><br>CC: <i>users@openser.org</i><br>Subject: <i>Re: [Users] openser and jabber</i><br>Date: <i>Wed, 05 Oct 2005 11:32:24 +0300</i><br>>Hello,<br>><br>>On 10/05/05 00:52, Matt L. Zhu wrote:<br>><br>>>hi,<br>>><br>>>has anyone successfully implemented the jabber module for openser? <br>>>i keep on getting this error the moment i enable jabber.so in <br>>>openser.cfg<br>>><br>>>1(0) get_connection(): Inherited open database connections, this is <br>>>not a good idea<br>>>1(0) db_init(): Could not create a connection<br>>>1(0) ERROR: child_init(-2): Error while connecting database<br>>>1(0) init_mod_child(): Error while initializing module usrloc<br>><br>>the error comes from the usrloc module. OpenSER was unable to open <br>>the connection to MySQL server. Check the username, password, host <br>>and database you gave to db_url parameter of usrloc module.<br>><br>>Daniel<br>><br>>>1(0) ERROR: open_uac_fifo: init_child failed<br>>>1(0) starting fifo server failed<br>>>1(0) XJAB: Unloading module ...<br>>><br>>><br>>>without jabber, everything works fine, here are my parameters.<br>>>modparam("jabber", "db_url", "mysql://root:root@localhost/sip")<br>>>modparam("jabber", "jaddress", "192.168.0.4")<br>>>modparam("jabber", "jdomain", "192.168.0.4")<br>>>modparam("jabber", "proxy", "127.0.0.1:5060 localhost:5060")<br>>><br>>>please help<br>>><br>>><br>>><br>>>_______________________________________________<br>>>Users mailing list<br>>>Users@openser.org<br>>>http://openser.org/cgi-bin/mailman/listinfo/users<br>>><br></font></BLOCKQUOTE>
On 10/05/05 19:06, Matt L. Zhu wrote:
i think the issue is usrloc and jabber are trying to share the same connection and creating an error. if i use dbmode=0 for usrloc, the error goes away, but i want to use persist database for usrloc as well.. is there a workaround?
It should not be a problem to reuse the connection. You can try to have separate databases for usrloc and for jabber module. What version of jabber server do you use?
Daniel
[...]
i do use separate databases for xjab and usrloc, usrloc is openser, and jab using openser_jab. I am using the newest jive messenger server. but that shouldn't affect the openser starting up. This issue is really frustrating, and i have been looking all over the map for resolutions, and seems to me that everyone is either doing jabber with usrloc's db_mode = 0 or not having jabber at all.
<BLOCKQUOTE style='PADDING-LEFT: 5px; MARGIN-LEFT: 5px; BORDER-LEFT: #A0C6E5 2px solid; MARGIN-RIGHT: 0px'><font style='FONT-SIZE:11px;FONT-FAMILY:tahoma,sans-serif'><hr color=#A0C6E5 size=1> From: <i>Daniel-Constantin Mierla <daniel@voice-system.ro></i><br>Reply-To: <i>daniel@voice-system.ro</i><br>To: <i>"Matt L. Zhu" <coder0000@hotmail.com></i><br>CC: <i>users@openser.org</i><br>Subject: <i>Re: [Users] openser and jabber</i><br>Date: <i>Thu, 06 Oct 2005 14:51:20 +0300</i><br>><br>><br>>On 10/05/05 19:06, Matt L. Zhu wrote:<br>><br>>>i think the issue is usrloc and jabber are trying to share the same <br>>>connection and creating an error. if i use dbmode=0 for usrloc, the <br>>>error goes away, but i want to use persist database for usrloc as <br>>>well.. is there a workaround?<br>><br>>It should not be a problem to reuse the connection. You can try to <br>>have separate databases for usrloc and for jabber module. What <br>>version of jabber server do you use?<br>><br>>Daniel<br>><br>>>[...]<br>><br></font></BLOCKQUOTE>