August 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

[freeswitch-users] play_and_get_digits is not blocking
by Paulo Vicentini 10 Aug '10

10 Aug '10

Hi, With ESL: this->execute ( 'play_and_get_digits',... $var .. $digits = $this->api_uuid_getvar ( $uuid, $var ); But the problem is that play_and_get_digits *is not blocking* so that I can't get dtmf digits I made a simple test by sleeping a few seconds after calling play_and_get_digits so that I was able to get digits...Is play_and_get_digits supposed to be a blocking blocking instruction? (FreeSWITCH Version 1.0.head (git-c3d6c64 2010-07-05 10-08-30 -0500)) Thank youPaulo

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Re: [Freeswitch-users] FS processing 302
by Kristian Kielhofner 10 Aug '10

10 Aug '10

No, it "just worked". -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: 'FreeSWITCH Users Help' <freeswitch-users(a)lists.freeswitch.org> Sent: Tue Aug 10 14:19:23 2010 Subject: Re: [Freeswitch-users] FS processing 302 Interesting that it worked for you. Currently my setup will not send a new invite to the maddr address in the 302 contact. Do you make any changes to get your setup to accept the 302 and send to the maddr address? -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Kristian Kielhofner Sent: Tuesday, August 10, 2010 11:35 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] FS processing 302 It's supposed to be but I've seen implementations use it when attempting to indicate a specific address to signal to. Freeswitch did "the right thing" and used the address from maddr in my tests so it should work for you. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Sent: Tue Aug 10 12:22:25 2010 Subject: Re: [Freeswitch-users] FS processing 302 Not sure... isn't that multicast? On Aug 10, 2010, at 11:05 AM, Chris Veazey wrote: > Freeswitch via the external profile(port 5080) is sending an Invite and > receiving a 302 response. > > Should Freeswitch by default be able to process the maddr of a 302 > redirect? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)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(a)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(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Different versions, different cpu load
by Juan Antonio Ibañez Santorum 10 Aug '10

10 Aug '10

Hello! I was been some test using one FS tarball version downloaded some weeks ago (FreeSWITCH Version 1.0.6 (svn-exported)) but it has some problems with odbc connections managemens. Now I'm using a git version (FreeSWITCH Version 1.0.head (git-b485f25 2010-07-30 19-46-05 -0400)) that seems to solve this problem but I noticed it has a high cpu load comparing with svn version. While I could manage more than 200 calls with a 75% CPU load into a dual core server using svn version, now, 50 calls consume this 75% cpu. I can see same modules are loaded (except new hash module needed in git version) and same scenario is used. I am not be able to find why now it uses more cpu than before. Any idea? Each testing call is a simple bridge to an external sip provider. Regards

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Leg_timeout question
by paul gore 10 Aug '10

10 Aug '10

Hi there, I am trying to accomplish a pretty simple thing in FS dialplan - ring one or multiple destinations with individual timeout per destination, if nobody answers - forward to a voicemail. Here is dial plan I use: <extension name="call_route_1688506814_16099860919" continue="false"><condition field="caller_id_number" expression="^(\d{7,20})$"/> <condition field="destination_number" expression="^(call_route_16099860919)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=false"/> <action application="set" data="ignore_early_media=true"/> <action application="sched_hangup" data="+7200 alloted_timeout"/> <action application="set" data="transfer_ringback=${us-ring}"/> <action application="set" data="ringback=${us-ring}"/> <action application="bridge" data="[leg_timeout=20]sofia/internal/1002%${domain_name},[leg_timeout=25]sofia/internal/1003%${domain_name}"/> <action application="sleep" data="500"/><action application="set" data="voicemail_greeting_number=1"/><action application="voicemail" data="default $${domain} 1002"/></condition></extension> What happens is incoming call from my provider rings ext 1002, but instead of going to the VM after 20 sec and no answer FS sends 602 (alloted_timeout) and the ring repeats again and so 3 times, after which call dies. But never goes to voice mail. What I am doing wrong?

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Re: [Freeswitch-users] Question about srtp secured B-Leg
by Kristian Kielhofner 10 Aug '10

10 Aug '10

It's probably not as clean as you'd like and there may be a better way to do it but why don't you just try calling the device twice, the first time with SRTP/SDES and the second (failover) time without? I'd think this would be compatible with the largest number of devices (whether using SRTP or not). -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Sent: Tue Aug 10 09:21:06 2010 Subject: [Freeswitch-users] Question about srtp secured B-Leg -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have late codec negotiation enabled and using Snom devices for encrypted calls (inernal only). When sdp of A-leg (Snom) has a=crypto I export sip_secure_media=true in my dialplan, so that B-leg (Snom) is also encrypted via srtp. That works nicely. When the B-Leg doesn't support srtp I got an "incompatible_device" error. This is also ok so far. Unfortunately this way you can't use Snom's optional srtp mechanism which consists simply of two media profiles in A-Leg's SDP (first RTP/SAVP then RTP/AVP) so the target can choose to encrypt or not. I don't know whether this is RFC conform or not nor if this makes really sense. I would like to be able to call a non encrypting ATA for checking for eaxample fax devices by simply calling them. My question is now, is there a way to tell FS sending outbound calls also with a non crypto offer within same sdp to B-Leg, so that it can choose to encrypt or not? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMYVJB4tZeNddg3dwRApBGAJ9d4keb5RQqMlS5sJ6jfN3VLAQkhwCeO3/x QlwcZMCW2ipsAIjV7cet2aU= =qzpR -----END PGP SIGNATURE----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Unexpected silence on the call
by VM Knott 10 Aug '10

10 Aug '10

Note: My skills with FS are mediocre at best, and my SIP knowledge is very green, so please bear with me. We have been experiencing an inconsistent problem with our FS conference server. On occassion, a person on the call will go silent. They remain connected to the call, but they have no voice (their channel does not appear muted in the CLI). They can hear everyone else on the call. If they hangup and reconnect to the call everything is okay. After performing some digging on a recent occurance of this behavior, it appeared to coincide with re-invites being sent from our platform back through the SIP Tandem. >From my understanding, the re-invite is initiated because of the session_expire parameter having been set to 15 minutes (1800 seconds). Our current assumption is that the 1800 seconds is expiring, and we are attempting to initiate the re-invite to the carrier, and the routing is not executing properly, perhaps? Has anyone experienced this problem? Is there a way to override the request to set the session_expire, or something? VMK

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Pb cleaning tree
by Jean-Yves F. Barbier 10 Aug '10

10 Aug '10

Hi list, I can't rebuild after a 'git pull' because of cleaning failure: make clean Making clean in src /bin/bash: line 26: /usr/bin/make: Liste d'arguments trop longue (means: arguments list too long) make: *** [clean-recursive] Erreur 1 How can I avoid that? JY -- I disagree with unanimity.

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collision risk with Debian iceweasel & icedove
by Jean-Yves F. Barbier 10 Aug '10

10 Aug '10

Hi list, FYI, if you use a Debian sid, there's a collision PB that prevents iceweasel and icedove to run on the same machine: > 13783 0.000076 open("/usr/local/freeswitch/lib/libnspr4.so", > O_RDONLY) = 32 > You have a libnspr4.so file getting in the way in your LD_LIBRARY_PATH. JY -- Very few profundities can be expressed in less than 80 characters.

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Question about srtp secured B-Leg
by Helmut Kuper 10 Aug '10

10 Aug '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I have late codec negotiation enabled and using Snom devices for encrypted calls (inernal only). When sdp of A-leg (Snom) has a=crypto I export sip_secure_media=true in my dialplan, so that B-leg (Snom) is also encrypted via srtp. That works nicely. When the B-Leg doesn't support srtp I got an "incompatible_device" error. This is also ok so far. Unfortunately this way you can't use Snom's optional srtp mechanism which consists simply of two media profiles in A-Leg's SDP (first RTP/SAVP then RTP/AVP) so the target can choose to encrypt or not. I don't know whether this is RFC conform or not nor if this makes really sense. I would like to be able to call a non encrypting ATA for checking for eaxample fax devices by simply calling them. My question is now, is there a way to tell FS sending outbound calls also with a non crypto offer within same sdp to B-Leg, so that it can choose to encrypt or not? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMYVJB4tZeNddg3dwRApBGAJ9d4keb5RQqMlS5sJ6jfN3VLAQkhwCeO3/x QlwcZMCW2ipsAIjV7cet2aU= =qzpR -----END PGP SIGNATURE-----

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handling of re-invite on b-leg
by Juan Backson 10 Aug '10

10 Aug '10

Hi, When freeswitch receives re-invite from the b-leg, is there any configuration in freeswitch that can tell freeswitch to simply proxy the re-invite back to the a-leg and not to enter the dialplan? Any help will be greatly appreciated. jb

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