Re: [Freeswitch-users] Proto specific hangup cause issue
That approach introduces a third party application to the setup (in order to capture and parse tha SIP messages) that adds a lot in terms of complexity and reliability ( and cpu usage). Also it could become a nightmare when you use a mix of protocols (iax, sip, h323) and technologies (openzap etc). In the case of a live debugging session, capturing is the most useful tool but if you want to troubleshoot based on historical data (CDRs) then you need some detailing. In addition you don't have to fill your databases with all the fields that FS gives you in an XML cdr. You could only pick those which are of interest in a particular application. Shelby Ramsey wrote: Hello, This is just my 2 cents ... but my experience has been that trying to catch all of the various variables (i.e. from XML_CDR) or otherwise can be a little trying (a row in your CDR database could be over 100 fields long!). The best option here is to catch the UUID's for the 2 call legs, capture all SIP messaging, parse and dump the messaging, and then correlate the calls from the CDR from there. Much easier than trying to do it from FS ... and most folks want to see SIP captures anyway (very broad set of tools to debug). Measuring things like ASR, PDD, etc in my opinion is much easier from the raw messaging than trying to do something with FS CDR records. On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We are currently in the migration process from our current system to a FS based setup. We are in the process of adapting our billing and routing to FS. All the CDRs (and variables) related issues that we have been discussing on this mailing list come from the need to extract the same level of information from FS as we do with our current closed source proprietary system. So, we chose FS because of the versatility it provides in every aspect (event handling, config implementation etc.) and we strongly believe that all these additions/fixes would be beneficial to many potential FS users. We are at your disposal for more details in case you need more information about what exactly we are trying to do. Basically, our approach is from the VoIP carrier's point of view rather than the PBX user's/implementor's. So, the details that we asked to be introduced to FS come from real life issues that we have faced during the last few years with various platforms and troubleshooting experiences with other VoIP carriers. Michael Collins wrote: Thanks for your feedback. It definitely helps to know not only what you need FS to do but why you need it to do so. Do you have FS in production right now? Just curious. Thanks, MC On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I already added 2 patches for you right. Just be clear about what you want. And I am grateful of that. it is protocol neutral, that's why it starts with sip_ I didn't know that. I thought that the sip_ variables are protocol specific. So one would expect there to be an iax_hangup_disposition, woomera_hangup_disposition etc? Maybe you should beat around the bush less with your requirements for your application you are expecting me to support for you. I am just trying to gather statistics for my providers as I would with any VoIP softswitch. (hangup causes per terminator per destination) I don't think that this is a specific application rather than a general necessity for VoIP carriers. It is also very useful for troubleshooting purposes : when I look at my CDRs to find a call that I got a complain for, I want to be able to tell if it was me or the provider who hanged up and gave a specific hangup cause, so that I can troubleshoot the issue better. Just be clear about what you want. I want FS to reach that level of detailing and maturity in all aspects so that it could be the softswitch of choice by any VoIP entrepreneur (or hobbyist) and it is my strong belief that this can only be done by the community giving feedback to the programmers about what they find useful or not (i.e. experience from real-life situations). The patches that you made the last few days were not intended for me exclusively but for anyone that will face the same situations using FS. If you want the community to stop sending feedback about features/improvements you may as well close down this mailing list or just use it as an announcement board. I wish I was a c programmer and get involved with the project actively. But I am not. And as far as I can tell most of the registered users in this list aren't either. So they only way we can help is by testing and suggesting. Anthony Minessale wrote: it is protocol
Re: [Freeswitch-users] Proto specific hangup cause issue
That approach introduces a third party application to the setup (in order to capture and parse tha SIP messages) that adds a lot in terms of complexity and reliability ( and cpu usage). Also it could become a nightmare when you use a mix of protocols (iax, sip, h323) and technologies (openzap etc). In the case of a live debugging session, capturing is the most useful tool but if you want to troubleshoot based on historical data (CDRs) then you need some detailing. In addition you don't have to fill your databases with all the fields that FS gives you in an XML cdr. You could only pick those which are of interest in a particular application. Shelby Ramsey wrote: Hello, This is just my 2 cents ... but my experience has been that trying to catch all of the various variables (i.e. from XML_CDR) or otherwise can be a little trying (a row in your CDR database could be over 100 fields long!). The best option here is to catch the UUID's for the 2 call legs, capture all SIP messaging, parse and dump the messaging, and then correlate the calls from the CDR from there. Much easier than trying to do it from FS ... and most folks want to see SIP captures anyway (very broad set of tools to debug). Measuring things like ASR, PDD, etc in my opinion is much easier from the raw messaging than trying to do something with FS CDR records. On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We are currently in the migration process from our current system to a FS based setup. We are in the process of adapting our billing and routing to FS. All the CDRs (and variables) related issues that we have been discussing on this mailing list come from the need to extract the same level of information from FS as we do with our current closed source proprietary system. So, we chose FS because of the versatility it provides in every aspect (event handling, config implementation etc.) and we strongly believe that all these additions/fixes would be beneficial to many potential FS users. We are at your disposal for more details in case you need more information about what exactly we are trying to do. Basically, our approach is from the VoIP carrier's point of view rather than the PBX user's/implementor's. So, the details that we asked to be introduced to FS come from real life issues that we have faced during the last few years with various platforms and troubleshooting experiences with other VoIP carriers. Michael Collins wrote: Thanks for your feedback. It definitely helps to know not only what you need FS to do but why you need it to do so. Do you have FS in production right now? Just curious. Thanks, MC On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I already added 2 patches for you right. Just be clear about what you want. And I am grateful of that. it is protocol neutral, that's why it starts with sip_ I didn't know that. I thought that the sip_ variables are protocol specific. So one would expect there to be an iax_hangup_disposition, woomera_hangup_disposition etc? Maybe you should beat around the bush less with your requirements for your application you are expecting me to support for you. I am just trying to gather statistics for my providers as I would with any VoIP softswitch. (hangup causes per terminator per destination) I don't think that this is a specific application rather than a general necessity for VoIP carriers. It is also very useful for troubleshooting purposes : when I look at my CDRs to find a call that I got a complain for, I want to be able to tell if it was me or the provider who hanged up and gave a specific hangup cause, so that I can troubleshoot the issue better. Just be clear about what you want. I want FS to reach that level of detailing and maturity in all aspects so that it could be the softswitch of choice by any VoIP entrepreneur (or hobbyist) and it is my strong belief that this can only be done by the community giving feedback to the programmers about what they find useful or not (i.e. experience from real-life situations). The patches that you made the last few days were not intended for me exclusively but for anyone that will face the same situations using FS. If you want the community to stop sending feedback about features/improvements you may as well close down this mailing list or just use it as an announcement board. I wish I was a c programmer and get involved with the project actively. But I am not. And as far as I can tell most of the registered users in this list aren't either. So they only way we can help is by testing and suggesting. Anthony Minessale wrote: it is protocol
Re: [Freeswitch-users] Proto specific hangup cause issue
That approach introduces a third party application to the setup (in order to capture and parse tha SIP messages) that adds a lot in terms of complexity and reliability ( and cpu usage). Also it could become a nightmare when you use a mix of protocols (iax, sip, h323) and technologies (openzap etc). In the case of a live debugging session, capturing is the most useful tool but if you want to troubleshoot based on historical data (CDRs) then you need some detailing. In addition you don't have to fill your databases with all the fields that FS gives you in an XML cdr. You could only pick those which are of interest in a particular application. Shelby Ramsey wrote: Hello, This is just my 2 cents ... but my experience has been that trying to catch all of the various variables (i.e. from XML_CDR) or otherwise can be a little trying (a row in your CDR database could be over 100 fields long!). The best option here is to catch the UUID's for the 2 call legs, capture all SIP messaging, parse and dump the messaging, and then correlate the calls from the CDR from there. Much easier than trying to do it from FS ... and most folks want to see SIP captures anyway (very broad set of tools to debug). Measuring things like ASR, PDD, etc in my opinion is much easier from the raw messaging than trying to do something with FS CDR records. On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: We are currently in the migration process from our current system to a FS based setup. We are in the process of adapting our billing and routing to FS. All the CDRs (and variables) related issues that we have been discussing on this mailing list come from the need to extract the same level of information from FS as we do with our current closed source proprietary system. So, we chose FS because of the versatility it provides in every aspect (event handling, config implementation etc.) and we strongly believe that all these additions/fixes would be beneficial to many potential FS users. We are at your disposal for more details in case you need more information about what exactly we are trying to do. Basically, our approach is from the VoIP carrier's point of view rather than the PBX user's/implementor's. So, the details that we asked to be introduced to FS come from real life issues that we have faced during the last few years with various platforms and troubleshooting experiences with other VoIP carriers. Michael Collins wrote: Thanks for your feedback. It definitely helps to know not only what you need FS to do but why you need it to do so. Do you have FS in production right now? Just curious. Thanks, MC On Tue, Dec 9, 2008 at 12:21 AM, Apostolos Pantsiopoulos [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I already added 2 patches for you right. Just be clear about what you want. And I am grateful of that. it is protocol neutral, that's why it starts with sip_ I didn't know that. I thought that the sip_ variables are protocol specific. So one would expect there to be an iax_hangup_disposition, woomera_hangup_disposition etc? Maybe you should beat around the bush less with your requirements for your application you are expecting me to support for you. I am just trying to gather statistics for my providers as I would with any VoIP softswitch. (hangup causes per terminator per destination) I don't think that this is a specific application rather than a general necessity for VoIP carriers. It is also very useful for troubleshooting purposes : when I look at my CDRs to find a call that I got a complain for, I want to be able to tell if it was me or the provider who hanged up and gave a specific hangup cause, so that I can troubleshoot the issue better. Just be clear about what you want. I want FS to reach that level of detailing and maturity in all aspects so that it could be the softswitch of choice by any VoIP entrepreneur (or hobbyist) and it is my strong belief that this can only be done by the community giving feedback to the programmers about what they find useful or not (i.e. experience from real-life situations). The patches that you made the last few days were not intended for me exclusively but for anyone that will face the same situations using FS. If you want the community to stop sending feedback about features/improvements you may as well close down this mailing list or just use it as an announcement board. I wish I was a c programmer and get involved with the project actively. But I am not. And as far as I can tell most of the registered users in this list aren't either. So they only way we can help is by testing and suggesting. Anthony Minessale wrote: it is protocol
[Freeswitch-users] How to get info from the b-leg
Hi, I am making a simple bridge between two call legs : Client --(a-leg)-- FS --(b-leg)--Provider How can I get information like network-address of the Provider, media-address, port used, media port used etc. from the second leg (b-leg)? Is all the information provided by the a-leg available for the b-leg as well? If, yes how can I access it? (and log it to my CDR file eventually) ___ 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
Re: [Freeswitch-users] How to get info from the b-leg
b-leg logging is enabled in the cdr module. but in the cdrs I cannot get
any variables that refer to the b-leg.
I tried the second way using ${sip_to_host} and {bleg_sip_to_host} but :
a) the variable returns the FS IP on the a-leg CDR (correctly)
b) the variable returns nothing on the b-leg CDR (empty). Shouldn't it
return the to host of the b-leg (my providers address)?
Anthony Minessale wrote:
2 options.
1) enable b-leg logging on the cdr module.
2) you can use the prefix bleg_ in a variable context to get to
caller_profile members
from the b leg.
eg ${bleg_caller_id_name}
On Wed, Dec 3, 2008 at 7:30 AM, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Hi,
I am making a simple bridge between two call legs :
Client --(a-leg)-- FS --(b-leg)--Provider
How can I get information like network-address of the Provider,
media-address,
port used, media port used etc. from the second leg (b-leg)?
Is all the information provided by the a-leg available for the
b-leg as
well? If, yese
how can I access it? (and log it to my CDR file eventually)
___
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ClueCon http://www.cluecon.com/
AIM: anthm
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FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED]
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Re: [Freeswitch-users] How to get info from the b-leg
I looked in the b-leg xml cdr and the ip address is not there (for
signaling) it is only there
for media (${remote_media_ip}) which is not the same thing now, is it?
While we are at it, I noticed that the ${local_media_port} and
${remote_media_port}
have the same value for each CDR (a or b leg). Shouldn't the first
variable hold the port
of the FS (on both legs) and the second variable the port of the client
(in the a-leg) or the port of
the provider (in the b-leg)?
Anthony Minessale wrote:
outgoing calls to not have an ip value set.
if you want to store the dest ip in the cdr you need to set it as a
custom variable and insert it
into your template for csv cdr or it will just be there in xml cdr
On Wed, Dec 3, 2008 at 8:18 AM, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
b-leg logging is enabled in the cdr module. but in the cdrs I
cannot get any variables that refer to the b-leg.
I tried the second way using ${sip_to_host} and {bleg_sip_to_host}
but :
a) the variable returns the FS IP on the a-leg CDR (correctly)
b) the variable returns nothing on the b-leg CDR (empty).
Shouldn't it return the to host of the b-leg (my providers address)?
Anthony Minessale wrote:
2 options.
1) enable b-leg logging on the cdr module.
2) you can use the prefix bleg_ in a variable context to get to
caller_profile members
from the b leg.
eg ${bleg_caller_id_name}
On Wed, Dec 3, 2008 at 7:30 AM, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
I am making a simple bridge between two call legs :
Client --(a-leg)-- FS --(b-leg)--Provider
How can I get information like network-address of the Provider,
media-address,
port used, media port used etc. from the second leg (b-leg)?
Is all the information provided by the a-leg available for
the b-leg as
well? If, yese
how can I access it? (and log it to my CDR file eventually)
___
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
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IRC: irc.freenode.net http://irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
iax:[EMAIL PROTECTED]/888
http://iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
pstn:213-799-1400
___ Freeswitch-users
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
IRC: irc.freenode.net http://irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
iax:[EMAIL PROTECTED]/888
http://iax:[EMAIL PROTECTED]/888
googletalk:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
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Re: [Freeswitch-users] How to get info from the b-leg
I'll try the patch. Thank you for your time.
As for the local and remote media ports :
I have an endpoint with IP xxx.xxx.xxx.xxx and an FS box with IP
yyy.yyy.yyy.yyy.
In a SIP bridge each side of the call leg between the two boxes will
pick a udp port in order to send/receive traffic.
In my CDRs (a-leg) when I call the ${remote_media_port} and
${local_media_port} it returns the same value (e.g. 18841) for both
endpoints
(yyy.yyy.yyy.yyy and xxx.xxx.xxx.xxx).
In my b-leg CDR (let's say yyy.yyy.yyy.yyy to zzz.zzz.zzz.zzz) both
variables hold the same value as well but a different
one than the a-leg's (e.g. 19871)
The way I thought it would happen is that each call leg would have a
pair of different port numbers for the two variables
because :
yyy would inform xxx that it should use port A
xxx would inform yyy that it should use port B
(that's one pair)
yyy would inform zzz that it should use port C
zzz would inform yyy that it should use port D
(that's another pair)
so for the a-leg : ${local_media_port} = A, ${remote_media_port} = B
for the b=leg : ${local_media_port} = C, ${remote_media_port} = D
Am I missing something?
Anthony Minessale wrote:
It's not an unreasonabe request so i added a patch you can test for me
to trunk that sets network_addr on the reciept of a reply to an invite
on an outbound call. and the 2 variables sip_reply_host and sip_reply_port
local and remote media port reflects the port being used between that
leg and it's remote connection eg the ip and port that the rtp stack
was asked to use.
On Wed, Dec 3, 2008 at 9:48 AM, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
I looked in the b-leg xml cdr and the ip address is not there (for
signaling) it is only there
for media (${remote_media_ip}) which is not the same thing now, is it?
While we are at it, I noticed that the ${local_media_port} and
${remote_media_port}
have the same value for each CDR (a or b leg). Shouldn't the first
variable hold the port
of the FS (on both legs) and the second variable the port of the
client (in the a-leg) or the port of
the provider (in the b-leg)?
Anthony Minessale wrote:
outgoing calls to not have an ip value set.
if you want to store the dest ip in the cdr you need to set it as
a custom variable and insert it
into your template for csv cdr or it will just be there in xml cdr
On Wed, Dec 3, 2008 at 8:18 AM, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
b-leg logging is enabled in the cdr module. but in the cdrs I
cannot get any variables that refer to the b-leg.
I tried the second way using ${sip_to_host} and
{bleg_sip_to_host} but :
a) the variable returns the FS IP on the a-leg CDR (correctly)
b) the variable returns nothing on the b-leg CDR (empty).
Shouldn't it return the to host of the b-leg (my providers
address)?
Anthony Minessale wrote:
2 options.
1) enable b-leg logging on the cdr module.
2) you can use the prefix bleg_ in a variable context to get
to caller_profile members
from the b leg.
eg ${bleg_caller_id_name}
On Wed, Dec 3, 2008 at 7:30 AM, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
I am making a simple bridge between two call legs :
Client --(a-leg)-- FS --(b-leg)--Provider
How can I get information like network-address of the
Provider,
media-address,
port used, media port used etc. from the second leg (b-leg)?
Is all the information provided by the a-leg available
for the b-leg as
well? If, yese
how can I access it? (and log it to my CDR file eventually)
___
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
IRC: irc.freenode.net http://irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
iax:[EMAIL PROTECTED]/888
http://iax:[EMAIL
[Freeswitch-users] Set variable for the outgoing leg
All the variables that I set show up only in the a-leg CDR. How can I set a variable that can be used during the b-leg CDR generation? ___ 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] mod_perl core dump
I have a perl script (for dialplan generation) that works fine. When I try to use the DBI module I get a segmentation fault. My OS is Linux CentOS 5.2 and I am using freeswitch-1.0.1. If I can recall correctly, some other guy had the same problem a few months ago but I cannot find the mailing list entry. Has anyone faced this problem besides me? ___ 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
Re: [Freeswitch-users] mod_perl multiple bindings
I tried using the $env object but it fails with :
2008-11-24 09:12:46 [ERR] mod_perl.c:69 Perl_safe_eval() [require
'/root/test_perl2.pl';]
Can't call method serialize on an undefined value at
/root/test_perl2.pl line 19.
Compilation failed in require at (eval 3) line 1.
The script I am using :
#!/usr/bin/perl
freeswitch::console_log(info, $env-serialize());
### MAIN START
$XML_STRING = '
?xml version=1.0 encoding=UTF-8 standalone=no?
document type=freeswitch/xml
section name=dialplan description=Perl RE Dial Plan For
FreeSwitch
context name=kinetix
extension name=test
condition field=destination_number expression=^.*$
action application=playback
data=tone_stream://path=${base_dir}/conf/tetris.ttml;loops=5/
/condition
/extension
/context
/section
/document
';
### MAIN END
1;
Anthony Minessale wrote:
currently it would entail binding everything to 1 script and detecting
which kind of section it
was in the script. Making it support many would require some coding
which would need to be done in every language module to keep them
uniform and we don't have the time for it right now. you can provide
a patch or post a bounty and maybe someone can work on it.
there is a magic env obj created in your script that has all the params
print $env-serialize();
print $env-serialize(xml);
$info = $env-getHeader(info);
On Thu, Nov 20, 2008 at 8:34 AM, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
What if I want to use one binding for directory, one for
configuration and one for dialplan?
While we are at it, how can I pass parameters so that I can fill
up my %XML_REQUEST when
the perl script is called from the xml dialplan? e.g. :
context name=route
extension name=default
condition field=destination_number
expression=^.*$
action application=perl
data=/root/test_perl2.pl /
/condition
/extension
/context
Anthony Minessale wrote:
no the languages only have one binding.
Do you really need more than one binding?
On Thu, Nov 20, 2008 at 6:20 AM, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi,
Is there a way to declare more than one script with its
binding in
perl.conf.xml?
Because from what I understood by reading the documentation,
is that
there are
no different sections to define different perl scripts with
bindings
like for example in the
xml_curl.conf.xml :
configuration name=xml_curl.conf description=cURL XML
Gateway
bindings
binding name=binding1
...
/binding
binding name=binding2
...
/binding
...
/bindings
/configuration
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Re: [Freeswitch-users] NoOp() equivalent?
Tried that, but the output of a simple action application=eval data=hello / does not appear in my console. I verified that the context that the eval is in gets executed. I have the loglevel set to debug in my switch.conf.xml by the way. Any help? Michael Collins wrote: Try eval http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eval -MC -Original Message- From: [EMAIL PROTECTED] [mailto:freeswitch- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 19, 2008 10:56 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] NoOp() equivalent? Is there an equivalent to asterisk's NoOp() so that I can print (within the XML dialplan) the stuff I want on the FS console? Like variables and stuff? ___ 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
Re: [Freeswitch-users] NoOp() equivalent?
Figured that out by myself. One has to raise the console debug output to debug. [EMAIL PROTECTED] wrote: Tried that, but the output of a simple action application=eval data=hello / does not appear in my console. I verified that the context that the eval is in gets executed. I have the loglevel set to debug in my switch.conf.xml by the way. Any help? Michael Collins wrote: Try eval http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eval -MC -Original Message- From: [EMAIL PROTECTED] [mailto:freeswitch- [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Wednesday, November 19, 2008 10:56 AM To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] NoOp() equivalent? Is there an equivalent to asterisk's NoOp() so that I can print (within the XML dialplan) the stuff I want on the FS console? Like variables and stuff? ___ 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 ___ 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] mod_perl multiple bindings
Hi, Is there a way to declare more than one script with its binding in perl.conf.xml? Because from what I understood by reading the documentation, is that there are no different sections to define different perl scripts with bindings like for example in the xml_curl.conf.xml : configuration name=xml_curl.conf description=cURL XML Gateway bindings binding name=binding1 ... /binding binding name=binding2 ... /binding ... /bindings /configuration ___ 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
Re: [Freeswitch-users] mod_perl multiple bindings
What if I want to use one binding for directory, one for configuration and one for dialplan? While we are at it, how can I pass parameters so that I can fill up my %XML_REQUEST when the perl script is called from the xml dialplan? e.g. : context name=route extension name=default condition field=destination_number expression=^.*$ action application=perl data=/root/test_perl2.pl / /condition /extension /context Anthony Minessale wrote: no the languages only have one binding. Do you really need more than one binding? On Thu, Nov 20, 2008 at 6:20 AM, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Is there a way to declare more than one script with its binding in perl.conf.xml? Because from what I understood by reading the documentation, is that there are no different sections to define different perl scripts with bindings like for example in the xml_curl.conf.xml : configuration name=xml_curl.conf description=cURL XML Gateway bindings binding name=binding1 ... /binding binding name=binding2 ... /binding ... /bindings /configuration ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 http://iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] pstn:213-799-1400 ___ 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
Re: [Freeswitch-users] mod_perl multiple bindings
That's great. And how do I pass the parameters when calling the script from my dialplan? e.g. action application=perl data=/root/test_perl.pl / Anthony Minessale wrote: currently it would entail binding everything to 1 script and detecting which kind of section it was in the script. Making it support many would require some coding which would need to be done in every language module to keep them uniform and we don't have the time for it right now. you can provide a patch or post a bounty and maybe someone can work on it. there is a magic env obj created in your script that has all the params print $env-serialize(); print $env-serialize(xml); $info = $env-getHeader(info); On Thu, Nov 20, 2008 at 8:34 AM, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: What if I want to use one binding for directory, one for configuration and one for dialplan? While we are at it, how can I pass parameters so that I can fill up my %XML_REQUEST when the perl script is called from the xml dialplan? e.g. : context name=route extension name=default condition field=destination_number expression=^.*$ action application=perl data=/root/test_perl2.pl / /condition /extension /context Anthony Minessale wrote: no the languages only have one binding. Do you really need more than one binding? On Thu, Nov 20, 2008 at 6:20 AM, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Is there a way to declare more than one script with its binding in perl.conf.xml? Because from what I understood by reading the documentation, is that there are no different sections to define different perl scripts with bindings like for example in the xml_curl.conf.xml : configuration name=xml_curl.conf description=cURL XML Gateway bindings binding name=binding1 ... /binding binding name=binding2 ... /binding ... /bindings /configuration ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 http://iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 http://iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] pstn:213-799-1400 ___ 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
Re: [Freeswitch-users] accountcode and user id not present in CDRs
Then what's the point of having it in the directory configuration file in
the first place, if you don't mind me asking?
I am really confused... :)
Anthony Minessale wrote:
you have to manually set the var on the channel in your dialplan.
On Tue, Nov 18, 2008 at 1:40 PM, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
The ${accountcode} variable IS set in the cdr_csv.xml conf file
yet the field is empty after the call.
Shouldn't it also show in the xml cdr? I thought the XML CDRs
included all of the session variables.
Brian West wrote:
Add ${accuntcode} to the CDR template in cdr.conf.xml... the template
can include any variables from the session.
/b
On Nov 18, 2008, at 12:50 PM, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
wrote:
Any help on how to define an endpoint (originating) and use some
attribute (like account_code or user id)
for billing purposes?
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Re: [Freeswitch-users] accountcode and user id not present in CDRs
No, this is an call FROM the user to another gateway.
The user whose data I want in my CDRs is the originator of the call.
Brian West wrote:
I'm going to guess that this is an inbound call to the user. Which
means the variables aren't set inbound to the user.
/b
On Nov 18, 2008, at 1:40 PM, [EMAIL PROTECTED] wrote:
The ${accountcode} variable IS set in the cdr_csv.xml conf file yet
the field is empty after the call.
Shouldn't it also show in the xml cdr? I thought the XML CDRs
included all of the session variables.
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[Freeswitch-users] NoOp() equivalent?
Is there an equivalent to asterisk's NoOp() so that I can print (within the XML dialplan) the stuff I want on the FS console? Like variables and stuff? ___ 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] accountcode and user id not present in CDRs
I am using acls (cidr) to accept incoming calls from a gateway that I do not want to register in my FS box. I have this gateway configured in a xml file : freeswitch/conf/directory/default/gateway1.xml include user id=GATEWAY1 mailbox= cidr=xxx.xxx.xxx.xxx/32 params param name=password value=1234/ /params variables variable name=accountcode value=CUSTOMER1/ variable name=user_context value=my_context/ variable name=effective_caller_id_name value=gateway1_callid/ variable name=effective_caller_id_number value=238383838383/ /variables /user /include I have the corresponding cidr in my ACL in acl.conf.xml. I am able to make a call from that gateway to my FS but in my CDRs (both xml or cdr_csv) the accountcode or user id is not present. Any help on how to define an endpoint (originating) and use some attribute (like account_code or user id) for billing purposes? ___ 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
Re: [Freeswitch-users] accountcode and user id not present in CDRs
The ${accountcode} variable IS set in the cdr_csv.xml conf file yet the
field is empty after the call.
Shouldn't it also show in the xml cdr? I thought the XML CDRs included
all of the session variables.
Brian West wrote:
Add ${accuntcode} to the CDR template in cdr.conf.xml... the template
can include any variables from the session.
/b
On Nov 18, 2008, at 12:50 PM, [EMAIL PROTECTED] wrote:
Any help on how to define an endpoint (originating) and use some
attribute (like account_code or user id)
for billing purposes?
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Re: [Freeswitch-users] FACILITY_NOT_SUBSCRIBED
It appears to have been cutoff. The last line that I see is: 2008-11-11 23:38:27 [NOTICE] switch_loadable_module.c:281 switch_loadable_module_process() Adding F Peter P GMX wrote: Aaargh, being able to read can be a real advantage sometimes. I have now put the log to http://pastebin.freeswitch.org/6098 Best regards Peter Brian West schrieb: If you look very close in the dialog box it says what they are. Its pastebin and freeswitch You failed the test :P /b On Nov 11, 2008, at 4:14 PM, Peter P GMX wrote: http://pastebin.freeswitch.org http://pastebin.freeswitch.orgasked for login credentials. Any idea where to get them from? I googled around, no solution fund. Wiki credentials don't work. Best regards Peter ___ 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
Re: [Freeswitch-users] FACILITY_NOT_SUBSCRIBED
Send a full debug from the FS console when a call is placed. That should give more of a clue as to where the issue is. Peter P GMX wrote: I receive a FACILITY_NOT_SUBSCRIBED message when I call from 1000 to 1001. I've read in the wiki: 50 FACILITY_NOT_SUBSCRIBED requested facility not subscribed [Q.850 This cause indicates that the user has requested a supplementary service, which is available, but the user is not authorized to use. I am wondering which supplementary service this could be. The invite message is as follows: == U xxx.xx.xx.186:2054 - xxx.xx.xxx.xx:5060 INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0. Via: SIP/2.0/UDP xxx.xx.xx.186:2054;branch=z9hG4bK-7qnms1c9aoqt;rport. From: Company1000 sip:[EMAIL PROTECTED];tag=pslbxvhxjo. To: sip:[EMAIL PROTECTED];user=phone. Call-ID: 3c267cac8c1f-smz6hpvv1e5h. CSeq: 2 INVITE. Max-Forwards: 70. Contact: sip:[EMAIL PROTECTED]:2054;line=rt3bxln1;flow-id=1. P-Key-Flags: keys=3. User-Agent: snom320/7.1.33. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Session-Expires: 3600;refresher=uas. Min-SE: 90. Proxy-Authorization: Digest username=1000,realm=xxx.xx.xxx.xx,nonce=201af2a0-af3f-11dd-af96-ebbe56552456,uri=sip:[EMAIL PROTECTED];user=phone,qop=auth,nc=0001,cnonce=0f67abbc,response=d984661b55085f64e3e6f93a09762eca,algorithm=MD5. Content-Type: application/sdp. Content-Length: 370. . v=0. o=root 1618416056 1618416056 IN IP4 xxx.xx.xx.186. s=call. c=IN IP4 xxx.xx.xx.186. t=0 0. m=audio 12472 RTP/AVP 8 0 9 2 3 18 4 101. a=rtpmap:8 pcma/8000. a=rtpmap:0 pcmu/8000. a=rtpmap:9 g722/8000. a=rtpmap:2 g726-32/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 g729/8000. a=rtpmap:4 g723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. == So there is nothing special in the SDP. Both Snom 320 phones register on the right domain (here:IP). They both can call their mailbox etc. They are behind a NAT on different public IPs. However calling each other doesn't work. Sofia status profile internal API CALL [sofia(status profile internal)] output: = Nameinternal Domain Name N/A DBName sofia_reg_internal Pres Hosts DialplanXML Context public Challenge Realm auto_from RTP-IP xxx.xx.xxx.xx SIP-IP xxx.xx.xxx.xx URL sip:[EMAIL PROTECTED]:5060 BIND-URLsip:[EMAIL PROTECTED]:5060 HOLD-MUSIC local_stream://moh CODECS G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN_ENABLEDtrue STUN_AUTO_DISABLE false Registrations: = Call-ID:3c2679b578bd-8brbg608itvr User: [EMAIL PROTECTED] Contact:Company1000 sip:[EMAIL PROTECTED]:2054;line=145ehzt5 Agent: snom320/7.1.33 Status: Registered(UDP)(unknown) EXP(2008-11-10 16:45:40) Host: freeswitch Call-ID:3c2677883993-68zb6go2xoip User: [EMAIL PROTECTED] Contact:Company1001 sip:[EMAIL PROTECTED]:2054;line=hcg076gv Agent: snom320/7.1.33 Status: Registered(UDP)(unknown) EXP(2008-11-10 16:45:41) Host: freeswitch = What can I do? Best regards Peter ___ 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] Inbound calls question
Hi, How can my FS accept inbound SIP calls from other gateways without the need of a registration from their part? I only need to be able to accept inbound calls from specific gateway IPs. I tried creating my own profile and gateway but it fails : Error Creating SIP UA for profile: myprofile Can someone give me some first-step directions? ___ 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
Re: [Freeswitch-users] Inbound calls question
Is it compulsory that I use different ports for different profiles? What if I want to use the same ports for my authenticated users and the non-authenticated ones? Birgit Arkesteijn wrote: Hi, I have no idea what that error message means, sorry. However, we have a setup where we only accept SIP from a single source and use ACL (and not a sip gateway profile), see: http://wiki.freeswitch.org/wiki/Acl Note, the thing that tripped us up was that incoming SIP on port 5060 by default comes in on the 'internal' gateway, and 5080 as external. We switched the port numbers around. Hope this helps. Cheers, Birgit On 04/11/08 16:24, [EMAIL PROTECTED] wrote: Hi, How can my FS accept inbound SIP calls from other gateways without the need of a registration from their part? I only need to be able to accept inbound calls from specific gateway IPs. I tried creating my own profile and gateway but it fails : Error Creating SIP UA for profile: myprofile Can someone give me some first-step directions? ___ 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
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
sleep for a couple of seconds But then you could only insert 1800 cdrs per hour... If I was to insert 36000 cdrs per hour this means that I have to open parse close 10 files per second. Imagine the I/O penalty just for opening - closing the file. (the persing is the same for both situations) David Knell wrote: [EMAIL PROTECTED] wrote: Yes, the xml files give you tons of info... but isn't it a little insufficient - performance wise - to open and close so many files in such a little time. In a PBX environment that wouldn't be an issue but if we get to the small-voip-carrier level (some thousand cdrs per hour) that could slow things down considerably, wouldn't it? Not that you'd notice. We run XML CDR to database scripting on each box that we use for switching, and it's a pretty trivial task compared with switching all that media. Doing it this way is:- (a) distributed - one process per box scales nicely; (b) robust - script down, DB down, no problem: files just queue up; (c) simple - the script logic is trivial: - while 1 - for each file in the XML CDR directory - open it - parse it (XML::Simple for us) - insert it in to the DB - delete it - sleep for a couple of seconds Two error cases: can't parse or can't find data which should be there: move the file in to another directory to be examined by real eyes; DB insert fails: break out of inner loop and it'll be retried after a short pause. --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
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
I'll try some tests with various combinations first and then decide what's best. I have to say that I was astonished to find out that freeswitch had so many event handlers from the very beginning. However it would be great if freeswitch had the options for extra functionality (auto log rotation, db cdrs etc) so that it meets the peculiarities of every different project. That would make a big difference compared to other softswitch solutions where the lack of such features prohibits people from using them (especially in the carrier grade level). User feedback (and wish lists) is the key for the success of an open source (or not) project... Thank you all for your replies. You' ve been very helpful. David Knell wrote: The sleep's done each time the directory's empty, not each time a file's written. File open and close are trivial (it's probably still cached), and the contents are going to have to be parsed wherever you process it. We've used exactly this to process deliver CDRs on boxes handling in excess of 500K mins/day without issue. And, looking at one now, the CDR processor's used about 4% of the CPU time of FreeSWITCH, and about half that of the MySQL database which it writes the records to, also on the local machine, from which they're simply copied to the main CDR processor. It performance simply isn't worth worrying about. --Dave sleep for a couple of seconds But then you could only insert 1800 cdrs per hour... If I was to insert 36000 cdrs per hour this means that I have to open parse close 10 files per second. Imagine the I/O penalty just for opening - closing the file. (the persing is the same for both situations) David Knell wrote: [EMAIL PROTECTED] wrote: Yes, the xml files give you tons of info... but isn't it a little insufficient - performance wise - to open and close so many files in such a little time. In a PBX environment that wouldn't be an issue but if we get to the small-voip-carrier level (some thousand cdrs per hour) that could slow things down considerably, wouldn't it? Not that you'd notice. We run XML CDR to database scripting on each box that we use for switching, and it's a pretty trivial task compared with switching all that media. Doing it this way is:- (a) distributed - one process per box scales nicely; (b) robust - script down, DB down, no problem: files just queue up; (c) simple - the script logic is trivial: - while 1 - for each file in the XML CDR directory - open it - parse it (XML::Simple for us) - insert it in to the DB - delete it - sleep for a couple of seconds Two error cases: can't parse or can't find data which should be there: move the file in to another directory to be examined by real eyes; DB insert fails: break out of inner loop and it'll be retried after a short pause. --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 -- David Knell, Director, 3C Limited T: 020 8114 8901 F: 020 3002 7257 M: 001 415 630 3031 http://www.3c.co.uk ___ 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
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Don't get me wrong. I am not trying to undermine the excellent work that has already been done on the event handling modules of FS. I myself am a big fun of modular constructs in order to achieve something complicated (e.g. the UNIX way). But there are some situations that we will always need some extra features in order to accomplish our goal. As I wrote before, my current system is using the cdr-push (from the gateway side) method to gather my cdrs. Many billing systems and many terminating providers are satisfied with that model alone. I am not. In the end of each hour I am gathering all the cdrs and checking them one by one against my database in order to verify that ALL of my cdrs were handled by my radius servers. In my future system (where a more batch-like mode is preferable) I am forced to use a cdr-pull method. My billing system will be responisble for pulling the cdrs from the gateways and then process them. So if we have a look at the event handling weaponry of FS the following modules cannot meet me needs : mod_event_multicast mod_event socket mod_radius_cdr mod_xmpp_event because they all rely on a mechanism where the gateway is pushing my cdrs to my billing system (and you need a checking mechanism to verify that all the cdrs were handled). So now, lets have a look at the alternatives : mod_xml_cdr : this was my first choice as it had all the info needed (even more). But the cdr per file approach proved out to be inadequate in terms of performance. I wrote a few lines of code (in perl) that did a listing of the directory, parsed the cdrs using XML::Simple (without even doing anything meaningful, e.g. checks, inserts to a database etc.) and I was not able to make my system parse more than a 30 cdrs/sec (reference 1000 USD average system, not drawing any conclusions, don't get me wrong). Also while the parser was running I had a high cpu utilization (as expected) that was competing with my FS service. Wouldn't that make a bad impact of my freeswitch performance? so I went to a different module mod_cdr_csv : the text .csv format with the plethora of attributes was perfect for my application but it would be a daunting task for me to make a script tha purges the Master csv while it was being used by FS. I googled the issue and found about the HUP method and the rotate directive in the config. So far so good. But it lacked some functionality. All I was trying to state in my previous messages was that it would be nice if the rotation was being initiated by FS, and maybe have different behavior depending on another directive or something. Let me explain what I mean. There are some people who don't care about what cdrs exist in a given rotated log file, as long as a) no cdrs get lost in the process of rotation , b) no duplicates exist. I can see that the HUP method satisfied this need. There is another group of people that DO care about what cdrs are in a rotated log file. E.g. : The file with the name Master.csv.2008-01-01-09-00-00 would only contain cdrs that were terminated from 2008-01-01 09:00:00 to 2008-01-01 09:59:59 if an hourly setup was desirable or from 2008-01-01 09:00:00 to 2008-01-01 09:04:59 if a five-minutes setup was desirable. That need is not covered by the current HUP method because some cdrs might escape from the next to the previous file due to the fact that there is a delay between cron executing the HUP and FS doing the file log rotation. In simple words : I am getting my job done with FS they way it already is (no question about that). BUT, I (or someone with different needs) could do his/her job better if some minor features were present. I am not saying that you should should embrace modules that meet the average needs yet sacrifice flexibility. I am merely suggesting extending the flexibility of the already existing ones. Put some more lego tiles in your box set :) Michael Collins wrote: /me sends Anthony’s post to the printer to be laminated and framed… J *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Anthony Minessale *Sent:* Thursday, October 30, 2008 6:10 AM *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] mod_cdr revival (or new module maybe) Actually, The goal is to not limit the functionality by over thinking how things will be used but to provide building blocks to make much more possible. Our analogy for this is that if you take the common lego reference, If you have something cool built out of legos but they are super glued together it limits what you *could* have done had then not been. The system that David described is indeed ideal. I have mentioned it more than once and its no coincidence that FreeSWITCH plays into that model to a tee. I do not mention it very often because I think understanding that concept alone is valuable advice that I'd prefer
[Freeswitch-users] mod_cdr revival (or new module maybe)
Hi, I saw in the wiki that the mod_cdr module is now unsupported. There is also a note about a revival of the module. I would like to ask the following : What is the current state of the revival process? (should we expect something in the near future?) Will it have the same functionality as before (DB support for instance)? Are there any plans for a brand new database specific event handler module? It would be great if there was one so that developers (especially those who develop billing applications) would not have to create their own hacks (cron scripts etc.) Thank you for your time, -- --- Apostolos Pantsiopoulos Kinetix Tele.com Support Center ___ 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
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). Michael Jerris wrote: Unsure at this time. There has been some work on mod_cdr_odbc. We generally advise against direct to db cdr methods without a very robust backup method for when the db is down. On Oct 29, 2008, at 9:57 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I saw in the wiki that the mod_cdr module is now unsupported. There is also a note about a revival of the module. I would like to ask the following : What is the current state of the revival process? (should we expect something in the near future?) Will it have the same functionality as before (DB support for instance)? Are there any plans for a brand new database specific event handler module? It would be great if there was one so that developers (especially those who develop billing applications) would not have to create their own hacks (cron scripts etc.) Thank you for your time, -- --- Apostolos Pantsiopoulos Kinetix Tele.com Support Center ___ 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
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). Michael Jerris wrote: Unsure at this time. There has been some work on mod_cdr_odbc. We generally advise against direct to db cdr methods without a very robust backup method for when the db is down. On Oct 29, 2008, at 9:57 AM, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hi, I saw in the wiki that the mod_cdr module is now unsupported. There is also a note about a revival of the module. I would like to ask the following : What is the current state of the revival process? (should we expect something in the near future?) Will it have the same functionality as before (DB support for instance)? Are there any plans for a brand new database specific event handler module? It would be great if there was one so that developers (especially those who develop billing applications) would not have to create their own hacks (cron scripts etc.) Thank you for your time, -- --- Apostolos Pantsiopoulos Kinetix Tele.com Support Center ___ 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
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Yes, the xml files give you tons of info... but isn't it a little insufficient - performance wise - to open and close so many files in such a little time. In a PBX environment that wouldn't be an issue but if we get to the small-voip-carrier level (some thousand cdrs per hour) that could slow things down considerably, wouldn't it? Thanks again for your prompt replies, Michael Collins wrote: Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). For the record, I've been dumping all my XML CDRs into a particular directory and letting a script pick them up and process them. I think this is the best of both worlds: you get individual files with tons of info on each call and you can have a process that picks up those files and inserts them into the db. If the db is down then the CDRs aren't lost - they just accumulate in the directory until you get the db/script thing working again. Just my $.02 -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 ___ 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
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
That's very good news. :) Shawn Lewis wrote: In regards to auto log rotation - YES YES ANTHM just completed that item for me, where by you can set the time in minutes i believe it was. I have not tested it yet, hope to this week. Shawn Michael Collins wrote: Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). For the record, I've been dumping all my XML CDRs into a particular directory and letting a script pick them up and process them. I think this is the best of both worlds: you get individual files with tons of info on each call and you can have a process that picks up those files and inserts them into the db. If the db is down then the CDRs aren't lost - they just accumulate in the directory until you get the db/script thing working again. Just my $.02 -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 ___ 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
Re: [Freeswitch-users] mod_cdr revival (or new module maybe)
Good point. I have got this kind of behavior (cdrs push model) in my current system (using radius servers). The only drawback of this method is that if you want to be absolutely sure that all the cdrs were handled by the web server (or radius server) you have to check at certain intervals every cdr one by one (and handle those left unhandled for various reasons (network, excessive web server load etc.)) But for my next project I am somewhat forced to use a cdrs-pull method where a process will pull cdrs from the server at its own pace making this extra check unnecessary. I will wait for an automatic log rotation as Shawn Lewis wrote. I think that will do the job. Michael Collins wrote: Yes, the xml files give you tons of info... but isn't it a little insufficient - performance wise - to open and close so many files in such a little time. In a PBX environment that wouldn't be an issue but if we get to the small-voip-carrier level (some thousand cdrs per hour) that could slow things down considerably, wouldn't it? Thanks again for your prompt replies, At that level of activity then I would assume you'd want a more robust solution which obviously would involve a server handling the CDRs separately. That's where XML is a real winner: it can POST CDRs to a web server and the webserver can handle all the pre-processing and db fun stuff. And if the connection to the webserver failed, the CDRs would be put on disk so that they aren't lost forever. Also, the webserver could cache the CDRs to its disk (or whatever storage) if the db itself went down but the webserver stayed up. Just a thought, anyway. It may be extra layers but it's also extra control. -MC Michael Collins wrote: Yes, I agree. But one could use the two methods combined (csv or xml + db) for redundancy. Is there any consideration regarding automatic log rotation (e.g. hourly, or user specified) without the need of a HUP? Now, that could make things a lot easier for the development of an external csv to db aggregation script because the script would read from a closed (not used by freeswitch at the time) CDRs file. And the developer could be sure that the cdrs contained in that file would have a hangup timestamp that could be described by the filename (e.g. 20080101_01.csv). For the record, I've been dumping all my XML CDRs into a particular directory and letting a script pick them up and process them. I think this is the best of both worlds: you get individual files with tons of info on each call and you can have a process that picks up those files and inserts them into the db. If the db is down then the CDRs aren't lost - they just accumulate in the directory until you get the db/script thing working again. Just my $.02 -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 ___ 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
Re: [Freeswitch-users] Voicemail Event
I think what he wants is to have some remote server get the events and the email so that users can access it from the remote server, not FS. He doesn't want it emailed to the end user. Anthony Minessale wrote: All of that information is already in the email =D you can template out the email with all that data which is expanded on the fly per message. by the time we finished adding what you want we will have recreated SMTP from scratch ;) On Wed, Oct 8, 2008 at 7:34 PM, Michael Jerris [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Voicemail metadata is already stored in a database (of your choice via odbc) and if you store the files on some remotely mountable location you should get the same effect. I'll try to throw an event in today but I think some of what your trying to do is already done for you. Mike On Oct 8, 2008, at 8:25 PM, Nicholas Amorim wrote: Yes, I can email them. But certainly would be more interesting to add a event to voicemail received. It opens a wide whole world of possiblities :P Including real-time alerts, etc. The info that I need: Which user received the voicemail Access to the file which voicemail was recorded Date/Time of received voicemail Just that, I guess. I would capture the event, send all those infos through an url and then delete the voicemail from the machine. The url receives it and stores on a database, making it kinda scalable. On Wed, Oct 8, 2008 at 7:38 PM, Anthony Minessale [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: you can email them ? On Wed, Oct 8, 2008 at 5:10 PM, Nicholas Amorim [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Deliver the vm message physically. ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 http://iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] pstn:213-799-1400 ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org mailto:Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] GTALK/JABBER/PAYPAL:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] IRC: irc.freenode.net http://irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] iax:[EMAIL PROTECTED]/888 http://iax:[EMAIL PROTECTED]/888 googletalk:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] pstn:213-799-1400 ___ 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
[Freeswitch-users] ODBC Dropping
I have ODBC going to mysql on the same box as FS and I keep having issues where it is dropping and then coming back. Any ideas where this might be occurring or how to fix it? FYI, the connection is broken long before I get the error messages below. I actually get those when the connection comes back up. Again, all of this is on the same box, so it isn't a lan/wan issue or anything like that. I suspect I will get a update to the latest, but I wanted to check if there are any known issues before I do that. [EMAIL PROTECTED] version FreeSWITCH Version 1.0.trunk (9577) [EMAIL PROTECTED] 2008-10-08 08:05:25 [CRIT] switch_odbc.c:248 db_is_up() The sql server is not responding for DSN freeswitch [STATE: 24000 CODE 0 ERROR: [unixODBC][MySQL][ODBC 3.51 Driver][mysqld-4.1.22-log]Invalid cursor state ] 2008-10-08 08:05:25 [INFO] switch_odbc.c:253 db_is_up() The connection has been re-established ___ 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
Re: [Freeswitch-users] IAX outbound trunk and distortion
No load as no other calls. Just one channel active. Sip to the same provider has worked previously. I recorded a test call on xlite if you want a copy of the wav? Cheers, Alex -- I sent this from my 3 mobile -- -original message- Subject: Re: [Freeswitch-users] IAX outbound trunk and distortion From: Brian West [EMAIL PROTECTED] Date: 26/09/2008 9:30 pm What kind of load and how many channels? /b On Sep 26, 2008, at 1:13 PM, Alex Kinch wrote: Hi, Just setup an IAX trunk from FS to a SIP provider who runs Asterisk, but getting a truckload of distortion when I make calls. The iax.conf.xml settings are as per the default. I've turned debugging on but just wondered where would be a good place to start trying to track down the problem? The codec appears to be PCMU/8000. Any suggestions welcome - this is the first time I've used IAX on FS, and have never had audio distortion issues with SIP trunks etc. Thanks, Alex ___ 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
Re: [Freeswitch-users] IAX outbound trunk and distortion
Sure, will send it over when I get back. Also appear to have a dialplan issue where a call on sip matches but an iax trunk doesn't, but I'll put that on a separate email. Alex -- I sent this from my 3 mobile -- -original message- Subject: Re: [Freeswitch-users] IAX outbound trunk and distortion From: Brian West [EMAIL PROTECTED] Date: 26/09/2008 10:06 pm if the recorded wav exhibits the issue sure. /b On Sep 26, 2008, at 3:39 PM, [EMAIL PROTECTED] wrote: No load as no other calls. Just one channel active. Sip to the same provider has worked previously. I recorded a test call on xlite if you want a copy of the wav? Cheers, Alex -- I sent this from my 3 mobile -- ___ 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
Re: [Freeswitch-users] Freeswitch with Audiocode Mediant 2000
it can be a codec issue , make sure to use g711 at both ends /updating latest firmware can help Original Message: - From: Gopal krishnan [EMAIL PROTECTED] Date: Tue, 23 Sep 2008 19:31:07 +0530 To: freeswitch-users@lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch with Audiocode Mediant 2000 Hi, I followed the below link to configure the Audiocode Mediant 2000 with Freeswitch http://wiki.freeswitch.org/index.php?title=Configuring_AudioCodes_MP-114/118 printable=yes but the above link is for FXO line, where I am using digital PRI line. when I try to dial I am getting call failed, the traffic from freeswitch were hitting audiocode the log as follows, attached with this email, *some sample SIP header as follows,* d:2h:17m:7s INVITE sip:[EMAIL PROTECTED][EMAIL PROTECTED]SIP/2.0 Via: SIP/2.0/UDP 172.20.176.31;rport;branch=z9hG4bKKmB9HrNr22HZQ Max-Forwards: 69 From: Extension 1002 sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;tag=j9a4e9Q4ycvtr To: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Call-ID: 7702517d-0413-122c-efab-0019d150d051 CSeq: 104969298 INVITE Contact: sip:[EMAIL PROTECTED]:5060 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9596M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 347 Remote-Party-ID: Extension 1002 sip:[EMAIL PROTECTED][EMAIL PROTECTED] ;screen=yes;privacy=off 1d:2h:17m:7s ( sip_stack)(212 ) ?? [WARNING] AcSIPParser: Unrecognized Header was detected at line: 12 1d:2h:46m:9s ( lgr_TrnkGrp)(344 ) !! [ERROR] #1:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number 9894929942 1d:2h:46m:9s ( lgr_psbrdif)(345 ) !! [ERROR] AcBoard::GetEndPoint- Can't find EndPoint for Dest:9894929942 Source:9894929942 SourceIp:ac14b01f 1d:2h:46m:9s ( lgr_psbrdif)(346 ) TrunkBoard::GetEndPoint- Current trunk status:0010 1d:2h:46m:9s ( lgr_call)(347 ) !! [ERROR] Call::GetEndPoint- Can't find endpoint for phone number 9894929942 *Freeswitch log* *as follows* http://pastebin.freeswitch.org/5635 So how to proceed in this stage. -- Thank you with regards, Gopal, myhosting.com - Premium Microsoft® Windows® and Linux web and application hosting - http://link.myhosting.com/myhosting ___ 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
Re: [Freeswitch-users] SCCP aka Skinny
There is no SCCP module for FS. CM only uses SCCP to talk to phones, it uses either MGCP or SIP to talk to gateways. So if you have a version that has SIP support (I believe 4.0), then you could connect CM to FS. Cavalera Claudio Luigi wrote: Hello, is there a way to interconnect fs to a Cisco Call Manager which is configured to speak SCCP protocol (aka Skinny) and not SIP? I did not found a mod_SCCP in the docs :-) Thanks, Claudio Internet Email Confidentiality Footer - La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. - ___ 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
Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch
We don't see much samples for PRI outbound dialing . India the line is Euro ISDN. Has anyone tested sangoma A101 cards with Openzap ? I am tring to build a front end web application , to dial using JS in FS which will dial a outbound no and bridge the call to the extension. Thank you Imthiyaz Original Message: - From: Martin Joseph [EMAIL PROTECTED] Date: Fri, 19 Sep 2008 09:20:45 -0700 To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch On Sep 19, 2008, at 6:23 AM, Gopal krishnan wrote: Hi, Basically I just want to test outbound alone with freeswitch, so I can use extensions.conf in the conf directory rite? -- I would forget about the asterisk dialplan then. It's very simple to configure an outbound SIP provider in the XML config for FS. Look here for setting up your outbound provider: http://wiki.freeswitch.org/wiki/Tested_Phone_Providers_Listing Look here for a simple example (look for dialplan): http://wiki.freeswitch.org/wiki/Home_PBX_Example I set up a simple outbound SIP tester from these two pages in very little time. Good luck, hope this helps, Marty ___ 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 mail2web.com - Microsoft® Exchange solutions from a leading provider - http://link.mail2web.com/Business/Exchange ___ 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] H.323 -dtmf-
Hi All, would FreeSWITCH 'transcode' H.245 alphanumeric DTMFs to an H.245 signal / rfc2833 H.323 device over G.729 codec ? Thanks for supporting, .TF ___ 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] SS7 and SIP
HI We want to try generate 5000 simultanious Voice broadcast calls . can the below config will work? SS7 Links Sangoma SMG---FreeSwitchBroadcasting Application ( SIP based) Thank you Imthiyaz mail2web.com What can On Demand Business Solutions do for you? http://link.mail2web.com/Business/SharePoint ___ 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
