Hi, My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so apologies if some of my terminology isn't quite correct. Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the card is green and wanrouter is installed and up in TDM_API mode, with the connection status showing as connected. Configured Openzap for 9 b and 1 d channel as described in Freeswitch Wiki. Then created a diaplan to fire off any calls preceded by 9 to the next available openzap channel. The problem I have is when I initiate an external call (using 9xxxxxxx) from an extension I can see Freeswitch allocating the call to the next available channel but then the just sits there and times out after 1 minute. With the cause stated as ORIGINATOR_CANCEL (guessing this is the time out) Here are the confs / dialplan as I have them: [span wanpipe] name => OpenZAP number => 1 trunk_type => e1 b-channel => 1:1-9 d-channel => 1:16 <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> <!--<param name="hold-music" value="$${moh_uri}"/>--> <!--<param name="enable-analog-option" value="call-swap"/>--> <!--<param name="enable-analog-option" value="3-way"/>--> </settings> <pri_spans> <span id="1"> <param name="mode" value="user"/> <param name="dialect" value="q931"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </pri_spans> </configuration> <include> <extension name="outgoing-pri"> <condition field="destination_number" expression="^9(\d{7,15})$"> <action application="bridge" data="openzap/1/a/$1"/> </condition> </extension> </include> Do the config files look ok? Is there anything else I should be configuring? Is there anything else I can use to debug or get information on the PRI status? Thanks in advance. Mark. Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). --------------------------------------------------------------------------------
My first post to the list. I’m a bit of a newb to FreeSwitch (and linux) so apologies if some of my terminology isn’t quite correct.
Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE?
Recently had a 9 channel ISDN30 (euro – q931) installed by BT (UK). We’ve hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the card is green and wanrouter is installed and up in TDM_API mode, with the connection status showing as connected. Configured Openzap for 9 b and 1 d channel as described in Freeswitch Wiki. Then created a diaplan to fire off any calls preceded by 9 to the next available openzap channel.
Looks good so far...
The problem I have is when I initiate an external call (using 9xxxxxxx) from an extension I can see Freeswitch allocating the call to the next available channel but then the just sits there and times out after 1 minute. With the cause stated as ORIGINATOR_CANCEL (guessing this is the time out)
okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has lots of useful information for how to gather log information, how to use the pastebin, etc. Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers: To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org). Welcome to the wonderful world of telephony debugging! -MC P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.)
I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! http://pastebin.freeswitch.org/7751 Will setup an IRC client and see if I can log onto the channel. Thanks again! -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 March 2009 16:50 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1
My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so apologies if some of my terminology isn't quite correct.
Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE?
Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the card is green and wanrouter is installed and up in TDM_API mode, with the connection status showing as connected. Configured Openzap for 9 b and 1 d channel as described in Freeswitch Wiki. Then created a diaplan to fire off any calls preceded by 9 to the next available openzap channel.
Looks good so far...
The problem I have is when I initiate an external call (using 9xxxxxxx) from an extension I can see Freeswitch allocating the call to the next available channel but then the just sits there and times out after 1 minute. With the cause stated as ORIGINATOR_CANCEL (guessing this is the time out)
okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has lots of useful information for how to gather log information, how to use the pastebin, etc. Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers: To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org). Welcome to the wonderful world of telephony debugging! -MC P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.) _______________________________________________ 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 Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). --------------------------------------------------------------------------------
Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room. So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be. Any ideas? Pastebin debug output is in my reply below. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 13 March 2009 14:16 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian). Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer! http://pastebin.freeswitch.org/7751 Will setup an IRC client and see if I can log onto the channel. Thanks again! -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 March 2009 16:50 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1
My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so apologies if some of my terminology isn't quite correct.
Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE?
Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the card is green and wanrouter is installed and up in TDM_API mode, with the connection status showing as connected. Configured Openzap for 9 b and 1 d channel as described in Freeswitch Wiki. Then created a diaplan to fire off any calls preceded by 9 to the next available openzap channel.
Looks good so far...
The problem I have is when I initiate an external call (using 9xxxxxxx) from an extension I can see Freeswitch allocating the call to the next available channel but then the just sits there and times out after 1 minute. With the cause stated as ORIGINATOR_CANCEL (guessing this is the time out)
okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs It has lots of useful information for how to gather log information, how to use the pastebin, etc. Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers: To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org). Welcome to the wonderful world of telephony debugging! -MC P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.) _______________________________________________ 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 Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). -------------------------------------------------------------------------------- _______________________________________________ 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
Any chance you can give one of us access to this system? Best thing to do would be to join #openzap on irc.freenode.net. -MC (IRC: mercutioviz) On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room.
So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be.
Any ideas? Pastebin debug output is in my reply below.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 13 March 2009 14:16 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1
I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian).
Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer!
http://pastebin.freeswitch.org/7751
Will setup an IRC client and see if I can log onto the channel.
Thanks again!
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 March 2009 16:50 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1
My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so apologies if some of my terminology isn't quite correct.
Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE?
Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the card is green and wanrouter is installed and up in TDM_API mode, with the connection status showing as connected. Configured Openzap for 9 b and 1 d channel as described in Freeswitch Wiki. Then created a diaplan to fire off any calls preceded by 9 to the next available openzap channel.
Looks good so far...
The problem I have is when I initiate an external call (using 9xxxxxxx) from an extension I can see Freeswitch allocating the call to the next available channel but then the just sits there and times out after 1 minute. With the cause stated as ORIGINATOR_CANCEL (guessing this is the time out)
okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs
It has lots of useful information for how to gather log information, how to use the pastebin, etc.
Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers:
To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org).
Welcome to the wonderful world of telephony debugging! -MC
P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.)
_______________________________________________ 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
Save paper - don't print this email unless you need to.
-------------------------------------------------------------------------------- NOTICE from RNID Typetalk
This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any).
--------------------------------------------------------------------------------
_______________________________________________ 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
_______________________________________________ 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
Not sure if I can give access to the system externally. I know our security policy doesn't allow for stuff like that though. I'll pop on to the IRC channel - thanks for the help so far, I'm really keen to get this working after tinkering for well over a week with it! Mark. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 16 March 2009 17:16 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Any chance you can give one of us access to this system? Best thing to do would be to join #openzap on irc.freenode.net. -MC (IRC: mercutioviz) On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room.
So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be.
Any ideas? Pastebin debug output is in my reply below.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 13 March 2009 14:16 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1
I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian).
Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer!
http://pastebin.freeswitch.org/7751
Will setup an IRC client and see if I can log onto the channel.
Thanks again!
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 March 2009 16:50 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1
My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so apologies if some of my terminology isn't quite correct.
Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE?
Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the card is green and wanrouter is installed and up in TDM_API mode, with the connection status showing as connected. Configured Openzap for 9 b and 1 d channel as described in Freeswitch Wiki. Then created a diaplan to fire off any calls preceded by 9 to the next available openzap channel.
Looks good so far...
The problem I have is when I initiate an external call (using 9xxxxxxx) from an extension I can see Freeswitch allocating the call to the next available channel but then the just sits there and times out after 1 minute. With the cause stated as ORIGINATOR_CANCEL (guessing this is the time out)
okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs
It has lots of useful information for how to gather log information, how to use the pastebin, etc.
Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers:
To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org).
Welcome to the wonderful world of telephony debugging! -MC
P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.)
_______________________________________________ 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
Save paper - don't print this email unless you need to.
-------------------------------------------------------------------------------- NOTICE from RNID Typetalk
This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any).
--------------------------------------------------------------------------------
_______________________________________________ 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
_______________________________________________ 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
_______________________________________________ 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
Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life. However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 17 March 2009 09:31 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Not sure if I can give access to the system externally. I know our security policy doesn't allow for stuff like that though. I'll pop on to the IRC channel - thanks for the help so far, I'm really keen to get this working after tinkering for well over a week with it! Mark. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 16 March 2009 17:16 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Any chance you can give one of us access to this system? Best thing to do would be to join #openzap on irc.freenode.net. -MC (IRC: mercutioviz) On Mon, Mar 16, 2009 at 9:53 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Quick update on this. We've had the Euro ISDN line checked by BT and it all checks out ok - engineers were able to originate and make calls into the equipment on the end of the line our comms room.
So, it looks like either Wanpipe / FS can't use the circuit but do report it as being up. Changed all the usual stuff like patch cables so I'm really at a dead end as to what this could be.
Any ideas? Pastebin debug output is in my reply below.
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 13 March 2009 14:16 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1
I've not used Asterisk or Yate before. I've picked this project up from another colleague who is on long term leave, but I know he did look at Asterisk before deciding FS was more suited to our requirements (replacement PBX for an ageing Meridian).
Thanks for the reply and pointers towards debugging. I've uploaded our output as directed from Openzap dumps plus the complete FS debug that appears when placing an outside call. Hopefully it can help to provide a possible answer!
http://pastebin.freeswitch.org/7751
Will setup an IRC client and see if I can log onto the channel.
Thanks again!
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 12 March 2009 16:50 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1
My first post to the list. I'm a bit of a newb to FreeSwitch (and linux) so apologies if some of my terminology isn't quite correct.
Welcome to FS! Just out of curiosity, have you ever used Asterisk or YATE?
Recently had a 9 channel ISDN30 (euro - q931) installed by BT (UK). We've hooked it up to our FreeSwitch setup with a Sangoma A101 card. Light on the card is green and wanrouter is installed and up in TDM_API mode, with the connection status showing as connected. Configured Openzap for 9 b and 1 d channel as described in Freeswitch Wiki. Then created a diaplan to fire off any calls preceded by 9 to the next available openzap channel.
Looks good so far...
The problem I have is when I initiate an external call (using 9xxxxxxx) from an extension I can see Freeswitch allocating the call to the next available channel but then the just sits there and times out after 1 minute. With the cause stated as ORIGINATOR_CANCEL (guessing this is the time out)
okay, some debugging info will be useful. Please read this wiki page first: http://wiki.freeswitch.org/wiki/Reporting_Bugs
It has lots of useful information for how to gather log information, how to use the pastebin, etc.
Specifically for this issue you'll need to use the pastebin because there will be so much information. Here are some pointers:
To see what's happening with openzap you'll need to use the "oz list" and "oz dump 1" at the command line (CLI). You'll also need to turn on debugging so that PRI messages show up. You'll need to capture the output on the CLI and put it into the pastebin. (http://pastebin.freeswitch.org).
Welcome to the wonderful world of telephony debugging! -MC
P.S. - We have a few IRC channels where you can join to get more real-time support: #freeswitch and #openzap on irc.freenode.net. (More details are in the wiki page I mentioned above.)
_______________________________________________ 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
Save paper - don't print this email unless you need to.
-------------------------------------------------------------------------------- NOTICE from RNID Typetalk
This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any).
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On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life.
Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed."
However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call.
Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC
We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things! Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819 http://pastebin.freeswitch.org/7819 I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818 http://pastebin.freeswitch.org/7818 Getting there (slowly) Mark. mercutioviz wrote:
On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life.
Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed."
However placing an external call into FS isn't as successful, looks like it can't assign a channel and terminates the call.
Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look.
-MC
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We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things! Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819 I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818 Getting there (slowly) Mark. <quote author="mercutioviz"> On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life.
Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed."
However placing an external call into FS isn't as successful, looks
like it can't assign a channel and terminates the call.
Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC _______________________________________________ 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 </quote> -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 March 2009 15:48 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Another update - this time (part) good news! Decided to run wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life.
Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed."
However placing an external call into FS isn't as successful, looks
like it can't assign a channel and terminates the call.
Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look. -MC _______________________________________________ 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 Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). --------------------------------------------------------------------------------
2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch there was a timer problem which was not solved yet. This caused channels to be busy in my case. I am not sure whether this is solved yet. Can anybody confirm? Best regards Peter Mark Tabron schrieb:
We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things!
Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819
I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818
Getting there (slowly)
Mark.
<quote author="mercutioviz"> On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Another update - this time (part) good news! Decided to run
wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life.
Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed."
However placing an external call into FS isn't as successful, looks
like it can't assign a channel and terminates the call.
Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look.
-MC
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</quote>
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 March 2009 15:48 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1
On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Another update - this time (part) good news! Decided to run
wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life.
Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed."
However placing an external call into FS isn't as successful, looks
like it can't assign a channel and terminates the call.
Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look.
-MC
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-------------------------------------------------------------------------------- NOTICE from RNID Typetalk
This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any).
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On Wed, Mar 18, 2009 at 10:18 AM, Peter P GMX <Prometheus001@gmx.net> wrote:
2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch there was a timer problem which was not solved yet. This caused channels to be busy in my case.
I am not sure whether this is solved yet. Can anybody confirm?
We're using ozmod_libpri which has it's own PRI handling. So far, so good... -MC
So the second issue is possibly known - really could do with a fix or a workaround for this as we plan to use E1's for all incoming traffic. Can anyone shed light on the first problem (extension rings for a fraction of a second then hangs up) I mentioned below, or is that possibly part of the same issue? -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Peter P GMX Sent: 18 March 2009 17:18 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 2 months ago when I struggled with E1 trunks and OpenZAP in freeswitch there was a timer problem which was not solved yet. This caused channels to be busy in my case. I am not sure whether this is solved yet. Can anybody confirm? Best regards Peter Mark Tabron schrieb:
We're a couple more steps forward from yesterday. Turned out some of my regex was incorrect, plus example #9 in the Freeswitch Dialplan Wiki has an extra space before one of the closing brackets in the default.xml example. After staring at the screen all day it's funny how you miss these things!
Situation now is I can get the call into FS but, it rings the extension for a fraction of a second then the call drops. Here's the contents of the public and default dialplans I'm using (as per example in the wiki) and the debug - http://pastebin.freeswitch.org/7819
I'm also seeing another issue when placing subsequent inbound calls, they bounce if hitting the same channel the first call came in to (typically /1:1). Again, grabbed a debug of this - http://pastebin.freeswitch.org/7818
Getting there (slowly)
Mark.
<quote author="mercutioviz"> On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Another update - this time (part) good news! Decided to run
wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life.
Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed."
However placing an external call into FS isn't as successful, looks
like it can't assign a channel and terminates the call.
Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look.
-MC
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</quote>
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 17 March 2009 15:48 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1
On Tue, Mar 17, 2009 at 4:24 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
Another update - this time (part) good news! Decided to run
wancfg_tdmapi again, using the same settings as we always did, and we can now make external calls. I suspect that whatever BT did yesterday kicked the circuit back into life.
Good. I can't tell you how many times I've spoken to a telco when there's a problem and the circuit magically comes back to life. They frequently claim, "We didn't do anything." I think that's a euphemism for "we did a reset and prayed."
However placing an external call into FS isn't as successful, looks
like it can't assign a channel and terminates the call.
Be sure that you have some routing mechanism in your public.xml file. Do you have a whole block of DID numbers? Anyway, pastebin your public.xml and a debug trace of an incoming call, including what phone number the caller dialed, and we'll take a look.
-MC
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Save paper - don't print this email unless you need to.
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On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
So the second issue is possibly known - really could do with a fix or a workaround for this as we plan to use E1's for all incoming traffic.
Can anyone shed light on the first problem (extension rings for a fraction of a second then hangs up) I mentioned below, or is that possibly part of the same issue?
I have experienced this before but I believe it was resolved by having the telco switch protocol dialects which is probably not an option for you. I think your best bet is to use ozmod_libpri and see if the issue is still present. -MC
Thanks, found an install guide on the FS Wiki for libpri - will get the server cloned then install and test. Shall report back. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 19 March 2009 16:12 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
So the second issue is possibly known - really could do with a fix or a workaround for this as we plan to use E1's for all incoming traffic.
Can anyone shed light on the first problem (extension rings for a fraction of a second then hangs up) I mentioned below, or is that possibly part of the same issue?
I have experienced this before but I believe it was resolved by having the telco switch protocol dialects which is probably not an option for you. I think your best bet is to use ozmod_libpri and see if the issue is still present. -MC _______________________________________________ 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 Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). --------------------------------------------------------------------------------
Installed libpri but I'm stuck on what entries to put in openzap.conf.xml, here's how I have the span setup at the moment: <libpri_spans> <span id="1"> <param name="node" value="cpe"/> <param name="switch" value="Q931"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </libpri_spans> Node and Switch type aren't documented for libpri from what I can tell - I know the former is either CPE or NET, though, I'm unsure what other values can be used for switch type. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Mark Tabron Sent: 19 March 2009 16:55 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Thanks, found an install guide on the FS Wiki for libpri - will get the server cloned then install and test. Shall report back. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: 19 March 2009 16:12 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 On Thu, Mar 19, 2009 at 2:08 AM, Mark Tabron <Mark.Tabron@rnid-typetalk.org.uk> wrote:
So the second issue is possibly known - really could do with a fix or a workaround for this as we plan to use E1's for all incoming traffic.
Can anyone shed light on the first problem (extension rings for a fraction of a second then hangs up) I mentioned below, or is that possibly part of the same issue?
I have experienced this before but I believe it was resolved by having the telco switch protocol dialects which is probably not an option for you. I think your best bet is to use ozmod_libpri and see if the issue is still present. -MC _______________________________________________ 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 Save paper - don't print this email unless you need to. ------------------------------------------------------------------------ -------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). ------------------------------------------------------------------------ -------- _______________________________________________ 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
Am Friday 20 March 2009 schrieb Mark Tabron:
Installed libpri but I'm stuck on what entries to put in openzap.conf.xml, here's how I have the span setup at the moment:
<libpri_spans> <span id="1"> <param name="node" value="cpe"/> <param name="switch" value="Q931"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </libpri_spans>
Node and Switch type aren't documented for libpri from what I can tell - I know the former is either CPE or NET, though, I'm unsure what other values can be used for switch type.
The value for switch is invalid, it's going to fall back to dms100 with that one set. Valid settings are: ni1, ni2, dms100, euroisdn, lucent5e, att4ess, gr303eoc and gr303tmc and "euroisdn" is the one you'll want for a E1 line. Another setting you may need is: <param name="l1" value="alaw"/> stkn -- ------------------------------------------------------------------------------- Stefan Knoblich | axsentis GmbH | Web: http://www.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | http://oss.axsentis.de/ Amtsgericht Koeln: HR B 56238 | Email: s.knoblich@axsentis.de UST-ID: DE244977565 | JID: s.knoblich@jabber.axsentis.de ------------------------------------------------------------------------------- Web: http://stkn.techmage.de/ Email: stkn@freeswitch.org IRC: #freeswitch-de @ irc.freenode.net
Installed libpri but I'm stuck on what entries to put in openzap.conf.xml, here's how I have the span setup at the moment:
<libpri_spans> <span id="1"> <param name="node" value="cpe"/> <param name="switch" value="Q931"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </libpri_spans>
Node and Switch type aren't documented for libpri from what I can tell
Thanks for the additional info. Found a problem with ozmod_libri.so missing from the mod directory after installation (fixed by manually copying file over) and I can now happily confirm I can make calls both in and out! Thanks for all the help on this - really appreciated. -----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Stefan Knoblich Sent: 20 March 2009 18:29 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Problem dialing out via E1 Am Friday 20 March 2009 schrieb Mark Tabron: -
I know the former is either CPE or NET, though, I'm unsure what other values can be used for switch type.
The value for switch is invalid, it's going to fall back to dms100 with that one set. Valid settings are: ni1, ni2, dms100, euroisdn, lucent5e, att4ess, gr303eoc and gr303tmc and "euroisdn" is the one you'll want for a E1 line. Another setting you may need is: <param name="l1" value="alaw"/> stkn -- ------------------------------------------------------------------------ ------- Stefan Knoblich | axsentis GmbH | Web: http://www.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | http://oss.axsentis.de/ Amtsgericht Koeln: HR B 56238 | Email: s.knoblich@axsentis.de UST-ID: DE244977565 | JID: s.knoblich@jabber.axsentis.de ------------------------------------------------------------------------ ------- Web: http://stkn.techmage.de/ Email: stkn@freeswitch.org IRC: #freeswitch-de @ irc.freenode.net _______________________________________________ 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 Save paper - don't print this email unless you need to. -------------------------------------------------------------------------------- NOTICE from RNID Typetalk This communication contains information which is confidential and may also be privileged. It is for the exclusive use of the addressee. If you are not the addressee, please note that any distribution, dissemination, copying or use of this communication or the information in it is prohibited. If you have received this message in error, please notify the sender immediately at the above e-mail address and delete the information from your computer system. Please note that neither RNID nor the sender accepts any responsibility for viruses and it is your responsibility to scan the email and attachments (if any). --------------------------------------------------------------------------------
participants (5)
-
Mark Tabron
-
MarkTab
-
Michael Collins
-
Peter P GMX
-
Stefan Knoblich