I've gotten FreeSWITCH running on this config: - Intel Atom D945GCLF in 1U rackmount - RedHat FC8 - Xorcom USB Astribank-8 (FXO) - Xorcom USB Astribank-8 (FXS) - Trunk FreeSWITCH code from a few days ago - zaptel-1.4.9.2.xpp.r5566 - Grandstream BT-100 SIP phones - Grandstream GXV-3000 SIP videophones - Various analog phone extensions First I tried installing on FC9. This didn't work because FC9 couldn't deal with the on-board NIC card on the D945GCLF. FC8 installed fine, though, so I just used that instead. Overall I am very impressed with FreeSWITCH. It is vastly easier to set up than Asterisk, and much cleaner. The XML config files are *so* much better than the ad hoc Asterisk config syntax. I like that I can make it run like a regular RedHat service, though I could not get it to run as user freeswitch -- it seems to want to run as root. I read through a bunch of the code and it is really well done. Thanks for a great open source project! Most things work: I set up the SIP phones and added the analog extensions, and can dial out on any phone over my POTS lines via Astribank FXO. I can receive incoming calls as well. I can videoconference between the videophones by using the code in the sample dialplan (default.xml) for "intercom" mode. Neat! Problems: Weirndess with FXS-originated calls ----------------------------------- If I call from an analog (FXS) extension to a BT-100, it rings normally. But if I then pick up the SIP phone, I can hear the ring on the analog extension get choppy. Soon after, the call is dropped. If I call from an analog (FXS) extension to a GXV-3000, it rings normally, but when I pick up the SIP phone, I get an immediate busy on the analog extension. If I call from an analog (FXS) extension to an X-Lite SoftPhone, it rings normally, but when I pick up the SIP phone, no audio comes through from the analog extension. If I call from either a BT-100, a GXV-3000, or an X-Lite SoftPhone *to* an analog (FXS) extension, it works fine. The dialplan is simple (note that I have 1- and 2-digit extensions): <!-- dial an OpenZAP channel number to get the corresponding analog extension --> <extension name="OpenZAP extensions"> <condition field="destination_number" expression="^(9|1[0-6])$"> <action application="set" data="dialed_ext=$1"/> <action application="bridge" data="OpenZAP/${dialed_ext}/1"/> </condition> </extension> <!-- dial a number in the directory to get the corresponding SIP extension; use video if possible --> <extension name="local-extension"> <condition field="destination_number" expression="^([3-6]\d)$"> <action application="set" data="dialed_ext=$1"/> <action application="export"><![CDATA[sip_h_Call-Info=<sip:$${domain}>;answer-after=0]]></action> <action application="export" data="sip_invite_params=intercom=true"/> <action application="export" data="sip_auto_answer=true"/> <action application="bridge" data="user/${dialed_ext}@$${domain}"/> </condition> </extension> In general, I found it a bit strange that you can't (it seems) put analog extensions into the directory. Is it true that you have to define analog extensions manually in the dialplan? That's kind of counterintuitive, especially for Asterisk users; the Asterisk Now GUI makes analog and SIP extensions look the same for all practical purposes. Echo ---- I was getting massive distortion on analog connections involving any GXV-3000 before I changed echo_cancel_level in /etc/openzap/zt.conf. The default value was 64. I changed it to 0 and the distortion went away, but of course I get lots of annoying echo now. Other SIP phones don't seem to have this problem. I have read in earlier posts to this list that the GXV-3000 has known problems. But the interesting thing is that it was working fine with Asterisk and the Astribanks (though on a different CPU & M/B). The Astribank seems to have its own EC, but I can't figure out how to turn it on. And I have no idea what to set the value in zt.conf to; 32 seems to work better than either 16 or 64 (!). I tried fxotune but that didn't do anything: it ran for a long time and generated an /etc/fxotone that seemed to have all zero values. Basically, it would be great to have an "echo cancellation HOWTO" for OpenZap users since EC seems to be offered in several different layers. I also notice a bunch of different EC modes in the OpenZap code -- does anyone know how to try different ones out? FAX detection ------------- This just doesn't seem to work. Here's the relevant code from my diaplan. (It looks a little odd because I send all unanswered calls from my POTS lines to the ext. 31 voicemail box.) <!-- ring all extensions for 20 seconds, then send to voicemail for extension 31 --> <!-- we answer immediately to prevent the alarm from emitting a fax/modem tone in some cases before a previous call has settled --> <extension name="incoming-astribank"> <condition field="destination_number" expression="^([1-4])$"> <action application="answer" /> <action application="tone_detect" data="fax 1100 r +5000 transfer fax XML default" /> <action application="set" data="dialed_ext=31"/> <action application="export" data="dialed_ext=31"/> <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="set" data="transfer_ringback=${us-ring}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/> <action application="set" data="call_timeout=20"/> <action application="sleep" data="1000"/> <action application="bridge" data="user/30@$${domain},user/31@$${domain},user/33@$${domain},user/36@$${domain},user/37@$${domain},OpenZAP/9/1,OpenZAP/10/1,OpenZAP/11/1"/> <action application="sleep" data="1000"/> <action application="voicemail" data="default $${domain} ${dialed_ext}"/> </condition> </extension> <!-- called when the tone_detect above triggers --> <extension name="fax"> <condition field="destination_number" expression="^fax$"> <action application="set" data="transfer_ringback=${us-ring}"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="call_timeout=20"/> <action application="bridge" data="OpenZAP/12/1"/> </condition> </extension> Thanks for any help with these issues. Dave
I would recommend you open a Jira on the audio issue. http://jira.freeswitch.org so we don't loose track of the issue. I know there is a problem with the astribank not hanging up properly. /b On Aug 12, 2008, at 5:31 PM, David Baggett wrote:
I've gotten FreeSWITCH running on this config:
- Intel Atom D945GCLF in 1U rackmount - RedHat FC8 - Xorcom USB Astribank-8 (FXO) - Xorcom USB Astribank-8 (FXS) - Trunk FreeSWITCH code from a few days ago - zaptel-1.4.9.2.xpp.r5566 - Grandstream BT-100 SIP phones - Grandstream GXV-3000 SIP videophones - Various analog phone extensions
First I tried installing on FC9. This didn't work because FC9 couldn't deal with the on-board NIC card on the D945GCLF. FC8 installed fine, though, so I just used that instead.
Overall I am very impressed with FreeSWITCH. It is vastly easier to set up than Asterisk, and much cleaner. The XML config files are *so* much better than the ad hoc Asterisk config syntax.
I like that I can make it run like a regular RedHat service, though I could not get it to run as user freeswitch -- it seems to want to run as root. I read through a bunch of the code and it is really well done. Thanks for a great open source project!
Most things work: I set up the SIP phones and added the analog extensions, and can dial out on any phone over my POTS lines via Astribank FXO. I can receive incoming calls as well. I can videoconference between the videophones by using the code in the sample dialplan (default.xml) for "intercom" mode. Neat!
Problems:
Weirndess with FXS-originated calls -----------------------------------
If I call from an analog (FXS) extension to a BT-100, it rings normally. But if I then pick up the SIP phone, I can hear the ring on the analog extension get choppy. Soon after, the call is dropped.
If I call from an analog (FXS) extension to a GXV-3000, it rings normally, but when I pick up the SIP phone, I get an immediate busy on the analog extension.
If I call from an analog (FXS) extension to an X-Lite SoftPhone, it rings normally, but when I pick up the SIP phone, no audio comes through from the analog extension.
If I call from either a BT-100, a GXV-3000, or an X-Lite SoftPhone *to* an analog (FXS) extension, it works fine.
The dialplan is simple (note that I have 1- and 2-digit extensions):
<!-- dial an OpenZAP channel number to get the corresponding analog extension -->
<extension name="OpenZAP extensions">
<condition field="destination_number" expression="^(9|1[0-6])$">
<action application="set" data="dialed_ext=$1"/>
<action application="bridge" data="OpenZAP/${dialed_ext}/1"/>
</condition>
</extension>
<!-- dial a number in the directory to get the corresponding SIP extension; use video if possible -->
<extension name="local-extension">
<condition field="destination_number" expression="^([3-6]\d)$">
<action application="set" data="dialed_ext=$1"/>
<action application="export"><![CDATA[sip_h_Call-Info=<sip:$$ {domain}>;answer-after=0]]></action>
<action application="export" data="sip_invite_params=intercom=true"/>
<action application="export" data="sip_auto_answer=true"/>
<action application="bridge" data="user/${dialed_ext}@$${domain}"/>
</condition>
</extension>
In general, I found it a bit strange that you can't (it seems) put analog extensions into the directory. Is it true that you have to define analog extensions manually in the dialplan? That's kind of counterintuitive, especially for Asterisk users; the Asterisk Now GUI makes analog and SIP extensions look the same for all practical purposes.
Echo ----
I was getting massive distortion on analog connections involving any GXV-3000 before I changed echo_cancel_level in /etc/openzap/zt.conf. The default value was 64. I changed it to 0 and the distortion went away, but of course I get lots of annoying echo now. Other SIP phones don't seem to have this problem. I have read in earlier posts to this list that the GXV-3000 has known problems. But the interesting thing is that it was working fine with Asterisk and the Astribanks (though on a different CPU & M/B).
The Astribank seems to have its own EC, but I can't figure out how to turn it on. And I have no idea what to set the value in zt.conf to; 32 seems to work better than either 16 or 64 (!).
I tried fxotune but that didn't do anything: it ran for a long time and generated an /etc/fxotone that seemed to have all zero values.
Basically, it would be great to have an "echo cancellation HOWTO" for OpenZap users since EC seems to be offered in several different layers. I also notice a bunch of different EC modes in the OpenZap code -- does anyone know how to try different ones out?
FAX detection -------------
This just doesn't seem to work. Here's the relevant code from my diaplan. (It looks a little odd because I send all unanswered calls from my POTS lines to the ext. 31 voicemail box.)
<!-- ring all extensions for 20 seconds, then send to voicemail for extension 31 -->
<!-- we answer immediately to prevent the alarm from emitting a fax/modem tone in some cases before a previous call has settled -->
<extension name="incoming-astribank">
<condition field="destination_number" expression="^([1-4])$">
<action application="answer" />
<action application="tone_detect" data="fax 1100 r +5000 transfer fax XML default" />
<action application="set" data="dialed_ext=31"/>
<action application="export" data="dialed_ext=31"/>
<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
<action application="bind_meta_app" data="2 b s record_session::$${base_dir}/recordings/${caller_id_number}.$ {strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
<action application="set" data="transfer_ringback=${us-ring}"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
<action application="set" data="call_timeout=20"/>
<action application="sleep" data="1000"/>
<action application="bridge" data="user/30@$${domain},user/31@$${domain},user/33@$${domain},user/ 36@$${domain},user/37@$${domain},OpenZAP/9/1,OpenZAP/10/1,OpenZAP/ 11/1"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="default $${domain} ${dialed_ext}"/>
</condition>
</extension>
<!-- called when the tone_detect above triggers -->
<extension name="fax">
<condition field="destination_number" expression="^fax$">
<action application="set" data="transfer_ringback=${us-ring}"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="call_timeout=20"/>
<action application="bridge" data="OpenZAP/12/1"/>
</condition>
</extension>
Thanks for any help with these issues.
Dave
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Brian West sip:brian@freeswitch.org
I had no problem with FXS to x-lite, do you have the latest SVN trunk or one of the tarballs? We may have fixed some issues if you have an older release. We have an application you can use in the dialplan called set_user for example put this as the very first extension in the dialplan <extension name="set_openzap_user" continue="true"> <condition field="source" expression="mod_openzap"/> <action application="set_user" data="${caller_id_number}@$${domain}"/> </condition> </extension> if you put that in an extension at the top of your dialplan just for calls from openzap then you will make that call assume the settings of the user in the directory with an id that matches the caller id num set in openzap then fall through to the rest of the dialplan. most things you *think* are not possible are just not forced into place =D as for astribank, I have heard there are a few issues with them and I do not have one to test them. The guy who supports them is around on IRC but he tends to favor the AST that's why he calls them ASTribank so we can try to work with him if he's willing. On Tue, Aug 12, 2008 at 5:31 PM, David Baggett <freeswitch@baggett.org>wrote:
I've gotten FreeSWITCH running on this config:
- Intel Atom D945GCLF in 1U rackmount - RedHat FC8 - Xorcom USB Astribank-8 (FXO) - Xorcom USB Astribank-8 (FXS) - Trunk FreeSWITCH code from a few days ago - zaptel-1.4.9.2.xpp.r5566 - Grandstream BT-100 SIP phones - Grandstream GXV-3000 SIP videophones - Various analog phone extensions
First I tried installing on FC9. This didn't work because FC9 couldn't deal with the on-board NIC card on the D945GCLF. FC8 installed fine, though, so I just used that instead.
Overall I am very impressed with FreeSWITCH. It is vastly easier to set up than Asterisk, and much cleaner. The XML config files are *so* much better than the ad hoc Asterisk config syntax.
I like that I can make it run like a regular RedHat service, though I could not get it to run as user freeswitch -- it seems to want to run as root. I read through a bunch of the code and it is really well done. Thanks for a great open source project!
Most things work: I set up the SIP phones and added the analog extensions, and can dial out on any phone over my POTS lines via Astribank FXO. I can receive incoming calls as well. I can videoconference between the videophones by using the code in the sample dialplan (default.xml) for "intercom" mode. Neat!
Problems:
Weirndess with FXS-originated calls -----------------------------------
If I call from an analog (FXS) extension to a BT-100, it rings normally. But if I then pick up the SIP phone, I can hear the ring on the analog extension get choppy. Soon after, the call is dropped.
If I call from an analog (FXS) extension to a GXV-3000, it rings normally, but when I pick up the SIP phone, I get an immediate busy on the analog extension.
If I call from an analog (FXS) extension to an X-Lite SoftPhone, it rings normally, but when I pick up the SIP phone, no audio comes through from the analog extension.
If I call from either a BT-100, a GXV-3000, or an X-Lite SoftPhone *to* an analog (FXS) extension, it works fine.
The dialplan is simple (note that I have 1- and 2-digit extensions):
<!-- dial an OpenZAP channel number to get the corresponding analog extension -->
<extension name="OpenZAP extensions">
<condition field="destination_number" expression="^(9|1[0-6])$">
<action application="set" data="dialed_ext=$1"/>
<action application="bridge" data="OpenZAP/${dialed_ext}/1"/>
</condition>
</extension>
<!-- dial a number in the directory to get the corresponding SIP extension; use video if possible -->
<extension name="local-extension">
<condition field="destination_number" expression="^([3-6]\d)$">
<action application="set" data="dialed_ext=$1"/>
<action
application="export"><![CDATA[sip_h_Call-Info=<sip:$${domain}>;answer-after=0]]></action>
<action application="export" data="sip_invite_params=intercom=true"/>
<action application="export" data="sip_auto_answer=true"/>
<action application="bridge" data="user/${dialed_ext}@$${domain}"/>
</condition>
</extension>
In general, I found it a bit strange that you can't (it seems) put analog extensions into the directory. Is it true that you have to define analog extensions manually in the dialplan? That's kind of counterintuitive, especially for Asterisk users; the Asterisk Now GUI makes analog and SIP extensions look the same for all practical purposes.
Echo ----
I was getting massive distortion on analog connections involving any GXV-3000 before I changed echo_cancel_level in /etc/openzap/zt.conf. The default value was 64. I changed it to 0 and the distortion went away, but of course I get lots of annoying echo now. Other SIP phones don't seem to have this problem. I have read in earlier posts to this list that the GXV-3000 has known problems. But the interesting thing is that it was working fine with Asterisk and the Astribanks (though on a different CPU & M/B).
The Astribank seems to have its own EC, but I can't figure out how to turn it on. And I have no idea what to set the value in zt.conf to; 32 seems to work better than either 16 or 64 (!).
I tried fxotune but that didn't do anything: it ran for a long time and generated an /etc/fxotone that seemed to have all zero values.
Basically, it would be great to have an "echo cancellation HOWTO" for OpenZap users since EC seems to be offered in several different layers. I also notice a bunch of different EC modes in the OpenZap code -- does anyone know how to try different ones out?
FAX detection -------------
This just doesn't seem to work. Here's the relevant code from my diaplan. (It looks a little odd because I send all unanswered calls from my POTS lines to the ext. 31 voicemail box.)
<!-- ring all extensions for 20 seconds, then send to voicemail for extension 31 -->
<!-- we answer immediately to prevent the alarm from emitting a fax/modem tone in some cases before a previous call has settled -->
<extension name="incoming-astribank">
<condition field="destination_number" expression="^([1-4])$">
<action application="answer" />
<action application="tone_detect" data="fax 1100 r +5000 transfer fax XML default" />
<action application="set" data="dialed_ext=31"/>
<action application="export" data="dialed_ext=31"/>
<action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/>
<action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>
<action application="set" data="transfer_ringback=${us-ring}"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>
<action application="set" data="call_timeout=20"/>
<action application="sleep" data="1000"/>
<action application="bridge" data="user/30@$${domain},user/31@$${domain},user/33@$${domain},user/36@ $${domain},user/37@$${domain},OpenZAP/9/1,OpenZAP/10/1,OpenZAP/11/1"/>
<action application="sleep" data="1000"/>
<action application="voicemail" data="default $${domain} ${dialed_ext}"/>
</condition>
</extension>
<!-- called when the tone_detect above triggers -->
<extension name="fax">
<condition field="destination_number" expression="^fax$">
<action application="set" data="transfer_ringback=${us-ring}"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set" data="call_timeout=20"/>
<action application="bridge" data="OpenZAP/12/1"/>
</condition>
</extension>
Thanks for any help with these issues.
Dave
_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
I used an SVN checkout from a few days ago -- Friday, I think. Thanks for the tip on the set_openzap_user app -- that's very cool. Despite my comment about the directory, In general I'm actually impressed with your foresight -- it's clear you've done a lot of the right things in abstracting away from the original Asterisk model. Should be a great platform for future ideas. Dave Anthony Minessale wrote:
I had no problem with FXS to x-lite, do you have the latest SVN trunk or one of the tarballs? We may have fixed some issues if you have an older release.
We have an application you can use in the dialplan called set_user
for example put this as the very first extension in the dialplan
<extension name="set_openzap_user" continue="true"> <condition field="source" expression="mod_openzap"/> <action application="set_user" data="${caller_id_number}@$${domain}"/> </condition> </extension>
if you put that in an extension at the top of your dialplan just for calls from openzap then you will make that call assume the settings of the user in the directory with an id that matches the caller id num set in openzap then fall through to the rest of the dialplan.
most things you *think* are not possible are just not forced into place =D
as for astribank, I have heard there are a few issues with them and I do not have one to test them. The guy who supports them is around on IRC but he tends to favor the AST that's why he calls them ASTribank so we can try to work with him if he's willing.
Ask Anthony how much time he and the other devs actually *thought* about what they were going to do *before* they started hacking code! The FS devs did more than just learn from Asterisk - they made practical application of those lessons. -MC Sent from my iPhone On Aug 12, 2008, at 6:00 PM, David Baggett <freeswitch@baggett.org> wrote:
I used an SVN checkout from a few days ago -- Friday, I think.
Thanks for the tip on the set_openzap_user app -- that's very cool.
Despite my comment about the directory, In general I'm actually impressed with your foresight -- it's clear you've done a lot of the right things in abstracting away from the original Asterisk model. Should be a great platform for future ideas.
Dave
Anthony Minessale wrote:
I had no problem with FXS to x-lite, do you have the latest SVN trunk or one of the tarballs? We may have fixed some issues if you have an older release.
We have an application you can use in the dialplan called set_user
for example put this as the very first extension in the dialplan
<extension name="set_openzap_user" continue="true"> <condition field="source" expression="mod_openzap"/> <action application="set_user" data="${caller_id_number}@$$ {domain}"/> </condition> </extension>
if you put that in an extension at the top of your dialplan just for calls from openzap then you will make that call assume the settings of the user in the directory with an id that matches the caller id num set in openzap then fall through to the rest of the dialplan.
most things you *think* are not possible are just not forced into place =D
as for astribank, I have heard there are a few issues with them and I do not have one to test them. The guy who supports them is around on IRC but he tends to favor the AST that's why he calls them ASTribank so we can try to work with him if he's willing.
_______________________________________________ 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
Feel free to join IRC during the day USA time and we can have a look at your openzap issues. We have a #freeswitch and a #openzap on irc.freenode.net On Tue, Aug 12, 2008 at 8:00 PM, David Baggett <freeswitch@baggett.org>wrote:
I used an SVN checkout from a few days ago -- Friday, I think.
Thanks for the tip on the set_openzap_user app -- that's very cool.
Despite my comment about the directory, In general I'm actually impressed with your foresight -- it's clear you've done a lot of the right things in abstracting away from the original Asterisk model. Should be a great platform for future ideas.
Dave
Anthony Minessale wrote:
I had no problem with FXS to x-lite, do you have the latest SVN trunk or one of the tarballs? We may have fixed some issues if you have an older release.
We have an application you can use in the dialplan called set_user
for example put this as the very first extension in the dialplan
<extension name="set_openzap_user" continue="true"> <condition field="source" expression="mod_openzap"/> <action application="set_user" data="${caller_id_number}@ $${domain}"/> </condition> </extension>
if you put that in an extension at the top of your dialplan just for calls from openzap then you will make that call assume the settings of the user in the directory with an id that matches the caller id num set in openzap then fall through to the rest of the dialplan.
most things you *think* are not possible are just not forced into place =D
as for astribank, I have heard there are a few issues with them and I do not have one to test them. The guy who supports them is around on IRC but he tends to favor the AST that's why he calls them ASTribank so we can try to work with him if he's willing.
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
In case it may benefit other Astribank users: One thing I had to do was "manually" download the firmware to the device on bootup. I've seen the devices get confused and lose their firmware, so I'm keeping an eye on that. In any case, I made a simple RedHat service wrapper that takes care of firmware reseting and uploading using the xpp_fxloader script that's part of the zaptel source. See below for the text of the script. It should go in /etc/init.d/astribank. You'll need to % chkconfig --add astribank as well. If you run it manually after bootup, you'll need to % service stop zaptel before doing so, and % service restart zaptel afterwards. Dave ----------------------------------------------------------------------- #!/bin/sh # # astribank This shell script takes care of loading firmware to \ # the Astribank USB Telephony interfaces # chkconfig: 2345 8 93 # description: The Astribank is a USB telephony device # config: /etc/sysconfig/astribank initdir=/etc/init.d RETVAL=0 case "$1" in start) # Load firmware to Astribank devices echo -n "Loading Astribank firmware: " /usr/share/zaptel/xpp_fxloader load RETVAL=$? exit $RETVAL ;; stop) # Unload Astribank hardware echo -n "Resetting Astribank firmware: " /usr/share/zaptel/xpp_fxloader reset RETVAL=$? exit $RETVAL ;; restart) $0 stop $0 start ;; *) echo "Usage: astribank {start|stop|restart}" exit 1 esac exit $RETVAL
participants (4)
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Anthony Minessale
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Brian West
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David Baggett
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Michael S Collins