Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
nonetheless, can you please see my bugnote in jira. ===== can you edit /usr/local/freeswitch/conf/autoload_configs/iax.conf.xml and set the debug param to 1. Then reproduce the case where it's not working. We can see if the mod_iax is even getting the digits. ===== also be sure to sync to the current release by entering "make current" from the build root. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ----- Original Message ---- From: Birgit Arkesteijn <birgit@westhawk.co.uk> To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, March 11, 2008 9:36:22 AM Subject: Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable Hi Anthony, I'm fairly sure that it's not an IAX problem. IAX2 messages are properly acked, sound (between IAX2 and SIP) is ok, etc, etc. The problem is (only) with DTMF, as far as I can tell. It's more likely to be in core or sofia_sip? Thanks, Birgit On 11/03/08 13:20, Anthony Minessale wrote:
Just to warn you we do not use mod_iax all that much. It was used as a tool to test the core of FreeSWITCH in the early stages of development and we just kinda left it hanging around since it does at least work. We had to add a bunch of code to the stock iaxclient lib to make it threadsafe etc because the lib is designed to handle 1 or 2 calls and we are using it in a b2bua type env where hundreds of concurrent calls may be possible. That said, since we more or less forked the code to avoid generating support issues for the other soft-phones that use iaxclient, you may want to check if there were any bugs in the real version of the library that match your issue.
http://sourceforge.net/projects/iaxclient
We've had little to no demand for IAX2 so we have not focused very much on it. We do have strategies for creating a more robust and perhaps even an "improved" implementation of the IAX2 protocol. Alas, it's pretty much purely an academic exercise because there is little reason to implement it.
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400
----- Original Message ---- From: Birgit Arkesteijn <birgit@westhawk.co.uk> To: Freeswitch-users@lists.freeswitch.org Sent: Monday, March 10, 2008 10:52:40 AM Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
Hi,
We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64).
I'm experiencing problems with DTMF: The IAX2 DTMF frames aren't always transferred into RTP packets. Sometimes none of the frames are translated, sometimes only the first one is omitted. All the IAX2 DTMF packets are ack-ed.
Our setup: PhoneFromHere.com <http://PhoneFromHere.com> applet (IAX2 protocol) -> FreeSWITCH (ns2.westhawk.co.uk <http://ns2.westhawk.co.uk>) -> SIP to SystemX (sofia/default/123@66.212.134.192)
I searched on jira.freeswitch.org <http://jira.freeswitch.org> for a bug that could explain this, but failed, so I submitted this as a bug, see http://jira.freeswitch.org/browse/FSCORE-109
Has anyone else experienced the same or a similar problem?
Thanks, Birgit
-- -- Birgit Arkesteijn, birgit@westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.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 ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ
Hi Anthony, Yes, I've done both and added comments & files to the bug report. Birgit On 12/03/08 13:17, Anthony Minessale wrote:
nonetheless, can you please see my bugnote in jira.
===== can you edit /usr/local/freeswitch/conf/autoload_configs/iax.conf.xml and set the debug param to 1. Then reproduce the case where it's not working. We can see if the mod_iax is even getting the digits. =====
also be sure to sync to the current release by entering "make current" from the build root.
Anthony Minessale II
----- Original Message ---- From: Birgit Arkesteijn <birgit@westhawk.co.uk> To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, March 11, 2008 9:36:22 AM Subject: Re: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
Hi Anthony,
I'm fairly sure that it's not an IAX problem. IAX2 messages are properly acked, sound (between IAX2 and SIP) is ok, etc, etc.
The problem is (only) with DTMF, as far as I can tell. It's more likely to be in core or sofia_sip?
Thanks, Birgit
On 11/03/08 13:20, Anthony Minessale wrote:
Just to warn you we do not use mod_iax all that much. It was used as a tool to test the core of FreeSWITCH in the early stages of development and we just kinda left it hanging around since it does at least work. We had to add a bunch of code to the stock iaxclient lib to make it threadsafe etc because the lib is designed to handle 1 or 2 calls and we are using it in a b2bua type env where hundreds of concurrent calls may be possible. That said, since we more or less forked the code to avoid generating support issues for the other soft-phones that use iaxclient, you may want to check if there were any bugs in the real version of the library that match your issue.
http://sourceforge.net/projects/iaxclient
We've had little to no demand for IAX2 so we have not focused very much on it. We do have strategies for creating a more robust and perhaps even an "improved" implementation of the IAX2 protocol. Alas, it's pretty much purely an academic exercise because there is little reason to implement it.
Anthony Minessale II
----- Original Message ---- From: Birgit Arkesteijn <birgit@westhawk.co.uk> To: Freeswitch-users@lists.freeswitch.org Sent: Monday, March 10, 2008 10:52:40 AM Subject: [Freeswitch-users] IAX2 DTMF -> SIP RTP not reliable
Hi,
We're running FreeSWITCH svn Trunk 7697 on SuSE Linux 9.3 (x86-64).
I'm experiencing problems with DTMF: The IAX2 DTMF frames aren't always transferred into RTP packets. Sometimes none of the frames are translated, sometimes only the first one is omitted. All the IAX2 DTMF packets are ack-ed.
Our setup: PhoneFromHere.com <http://PhoneFromHere.com> applet (IAX2 protocol) -> FreeSWITCH (ns2.westhawk.co.uk <http://ns2.westhawk.co.uk>) -> SIP to SystemX (sofia/default/123@66.212.134.192)
I searched on jira.freeswitch.org <http://jira.freeswitch.org> for a bug that could explain this, but failed, so I submitted this as a bug, see http://jira.freeswitch.org/browse/FSCORE-109
Has anyone else experienced the same or a similar problem?
Thanks, Birgit
-- -- Birgit Arkesteijn, birgit@westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>
I am using svn from 3/7/08. If enum returns anything longer than the following string, FS gets "No Match" iax/xxx:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@TLDD.NLD/12345678901 However, dig has no problem getting the full string (which in this case is another 5 characters. I guess there is a built in limit in FS for what enum returns? I would have setup a gateway for this but I cannot find any way to setup iax gateways. Is there any? -F
Can you provide an exact record I can lookup to replicate this issue? /b On Mar 12, 2008, at 1:42 PM, Frank - IMPACT wrote:
I am using svn from 3/7/08.
If enum returns anything longer than the following string, FS gets "No Match"
iax/xxx:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@TLDD.NLD/12345678901
However, dig has no problem getting the full string (which in this case is another 5 characters.
I guess there is a built in limit in FS for what enum returns?
I would have setup a gateway for this but I cannot find any way to setup iax gateways. Is there any?
-F
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No sorry. this is an internal enum server we use for LCR. However, here is the record from our enum dns server *.4.9.3.9.3.2.1 IN NAPTR 200 10 "u" "E2U+SIP" "!^\\+(.*)$!iax/XXX:YYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYY@test.voipjet.com/\\1!" . -----Original Message----- Can you provide an exact record I can lookup to replicate this issue? /b On Mar 12, 2008, at 1:42 PM, Frank - IMPACT wrote:
I am using svn from 3/7/08.
If enum returns anything longer than the following string, FS gets "No Match"
iax/xxx:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@TLDD.NLD/12345678901
However, dig has no problem getting the full string (which in this case is another 5 characters.
I guess there is a built in limit in FS for what enum returns?
I would have setup a gateway for this but I cannot find any way to setup iax gateways. Is there any?
-F
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You know I think I see your problem here... you're storing the credentials in ENUM... That's not exactly a good idea... Should someone gain access to that you're opening yourself to a HUGE Fraud impact (into the hundreds of thousands of dollars range in the course of a few days if voipjet in this case doesn't have good fraud controls in place)
From: Frank - IMPACT <frank@impactfax.com> Reply-To: <freeswitch-users@lists.freeswitch.org> Date: Wed, 12 Mar 2008 18:00:42 -0400 To: <freeswitch-users@lists.freeswitch.org> Subject: Re: [Freeswitch-users] enum return string length problem
No sorry. this is an internal enum server we use for LCR.
However, here is the record from our enum dns server
*.4.9.3.9.3.2.1 IN NAPTR 200 10 "u" "E2U+SIP" "!^\\+(.*)$!iax/XXX:YYYYYYYYYYYYYYYYYYYYYYYYYYYYYYYY@test.voipjet.com/\\1!" .
-----Original Message-----
Can you provide an exact record I can lookup to replicate this issue?
/b
On Mar 12, 2008, at 1:42 PM, Frank - IMPACT wrote:
I am using svn from 3/7/08.
If enum returns anything longer than the following string, FS gets "No Match"
iax/xxx:XXXXXXXXXXXXXXXXXXXXXXXXXXXXXXXX@TLDD.NLD/12345678901
However, dig has no problem getting the full string (which in this case is another 5 characters.
I guess there is a built in limit in FS for what enum returns?
I would have setup a gateway for this but I cannot find any way to setup iax gateways. Is there any?
-F
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when FS is started from a putty window and left in foreground, the output is nicely color coded. but if a remote connection is made with fs.pl, all the color coding output is lost. is there a way to get the color output with fs.pl ?
At a guess, it would need to be sending ANSI escape sequences... On Mon, 2008-03-17 at 12:06 -0400, Frank - IMPACT wrote:
when FS is started from a putty window and left in foreground, the output is nicely color coded. but if a remote connection is made with fs.pl, all the color coding output is lost.
is there a way to get the color output with fs.pl ?
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We do coloring based on log level, which should be easy enough to add support for in the perl script. Mike On Mar 17, 2008, at 2:52 PM, Daniel Swarbrick wrote:
At a guess, it would need to be sending ANSI escape sequences...
On Mon, 2008-03-17 at 12:06 -0400, Frank - IMPACT wrote:
when FS is started from a putty window and left in foreground, the output is nicely color coded. but if a remote connection is made with fs.pl, all the color coding output is lost.
is there a way to get the color output with fs.pl ?
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Have you looked at fsconsole.pl, in the same directory as fs.pl? It appears to have some colour coding in it, based on log level. On Mon, 2008-03-17 at 12:06 -0400, Frank - IMPACT wrote:
when FS is started from a putty window and left in foreground, the output is nicely color coded. but if a remote connection is made with fs.pl, all the color coding output is lost.
is there a way to get the color output with fs.pl ?
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I was using fsconsole.pl and after a short amount of output to the log window (anywhere from 10 - 100 lines), fsconsole crashes. It leaves the terminal in a strange state that requires a terminal "reset" from the command line (I am doing this from a putty window). Up until the ab-end of fsconsole, it seems to work fine. I typically start fsconsole, then run "log info" and then change to the log window. From there it crashes. I am using 9210 svn of freeswitch. Has anyone else ever seen fsconsole do this?
That program is out of date. The underlying poe engine has bugs, we need to contact the author of the module or rework it to use the FreeSWITCH::Client module that I wrote for it. On Tue, Sep 9, 2008 at 8:52 AM, Frank - IMPACT <frank@impactfax.com> wrote:
I was using fsconsole.pl and after a short amount of output to the log window (anywhere from 10 - 100 lines), fsconsole crashes. It leaves the terminal in a strange state that requires a terminal "reset" from the command line (I am doing this from a putty window).
Up until the ab-end of fsconsole, it seems to work fine.
I typically start fsconsole, then run "log info" and then change to the log window. From there it crashes.
I am using 9210 svn of freeswitch.
Has anyone else ever seen fsconsole do this?
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
participants (8)
-
Anthony Minessale
-
Anthony Minessale
-
Birgit Arkesteijn
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Brian West
-
Daniel Swarbrick
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Frank - IMPACT
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Ken Rice
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Michael Jerris