December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_easyroute and dialling via a gateway (not appending an IP to the return route)
by Gavin Henry 10 Dec '10

10 Dec '10

Hi all, With easyroute how do I get it *not* to stick the gateway IP at the end of THE ROUTE, i.e. to dial out of a gateway? I just want sofia/gateway/suretec/441224279484 returned for entries that *are not* in the database number table. It's fine to have @XX.XX.XX.XX for entries *in* the database. I've checked mod_easyroute.c and I don't think you can? [DEBUG] mod_easyroute.c:242 THE ROUTE [sofia/gateway/suretec/441224279484(a)XX.XX.XX.XX] My tables are like: freeswitch=# select * from gateways; gateway_id | gateway_ip | group | limit | techprofile ------------+-------------+----------+-------+------------------------ 0 | XX.XX.XX.XX | suretec | 300 | sofia/gateway/suretec (1 row) freeswitch=# select * from numbers; number_id | gateway_id | number | acctcode | translated -----------+------------+--------------+----------+------------ 0 | 0 | 441224279484 | 100 | (1 row) easyroute config:  <param name="default-techprofile" value="sofia/gateway/suretec"/>  <param name="default-gateway" value="suretec"/> FS CLI results: *In* the database (which is where I'm happy with an IP on the end as a client will be getting the DID/DDI delivered to their IP address): freeswitch@internal> easyroute 441224279484 Number Limit Group AcctCode Dialstring 441224279484 300 surevoip 100 sofia/gateway/suretec/441224279484(a)XX.XX.XX.XX *Not* in the database (which is where *I'm not* happy with an IP on the end as it needs to go through the gateway definition): freeswitch@internal> easyroute 441224279485 Number Limit Group AcctCode Dialstring 441224279485 9999 sofia/gateway/suretec/441224279485@suretec Thanks. -- http://www.suretecsystems.com/services/openldap/ http://www.surevoip.co.uk

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No sip BYE from gateway at hangup
by Fraser Redmond 10 Dec '10

10 Dec '10

I've got a gateway that seems to be sending the sip BYE really late. If the local (softphone) party hangs up then everything happens as normal as I'd expect it to. If the end terminated through the gateway hangs up then it's another 15-40 sec before Freeswitch finds out it's hung up and processes the end of the call. I did a "sofia profile external siptrace on" and during that 15-40 sec there was no activity until the end when the BYE turned up. I had the gateway run a sip trace their end and they said everything looked normal to them, and they said: "We have reviewed all of the calls specified both on the SIP signaling side and the SS7 side. Under certain circumstances, some calls can take longer to fully tear from the PSTN than others. " Which I interpret as a polite shrug of the shoulders :-) Any ideas of what I could look at? Or is it just one of those things that happens? The other gateways I have registered don't have the same problem. Cheers, Fraser

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Asterisk migrations
by Michael Collins 10 Dec '10

10 Dec '10

Hey gang, I was just curious if anyone out there had created any tools to assist with migrating from Asterisk (or Trixbox, etc.) to FreeSWITCH. We're starting to see this scenario more and I would like to find out how our community members are handling it. Thanks, MC

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Stop offering video on outbound if no video offered on inbound
by peely 10 Dec '10

10 Dec '10

Hi, I have a "registrar" sip profile which I have set H263 and H264 in the codec-prefs to allow video. Now, I see that when a call is offered which only provides audio codecs, FS still offers video codecs on the outbound call. Interestingly I have also occasionally seen this on session timer updates, where an audio codec had been negotiated but then video offered suddenly in the timer refresh, which has confused a few devices. Is there an option to not offer video codecs on the egress if no video codec is offered on the ingress, and possibly to allow audio transcoding but no video transcoding? I guess I could have a stab at a set of swanky dialplan entries but wondered if there's a specific profile option. Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Stop-offering-video-on-outbou… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Disabling Google Voice Voicemail
by mazilo 10 Dec '10

10 Dec '10

My FS hosted on a Seagate http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar DockStar works just fine to handle I/O calls with Google Voice (GV) service using mod_dingaling. However, there is a simple problem with the GV voicemail service that kicks in after two rings on my extension. unless someone knows this, there seems to be no way to turn off and/or disable GV voicemail. As such, I am looking for an alternate solution if anyone here has one to share. This way, I don't have to rush to my FS extension whenever it rings to avoid GV voicemail kicks in, let alone some of the calls from not on GV services through mod_dingaling. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Disabling-Google-Voice-Voicem… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket)
by Peter Olsson 10 Dec '10

10 Dec '10

No, and all timing seems ok (both good audio and timer_test returns with good results). And we're actually talking about 10 seconds here :) This is on Windows, and I'm starting with -hp, I wonder if I might have some kind of thread priority issue - I can't remember seeing this before adding -hp.. hmm, I will check that out. Peter Olsson ----- Reply message ----- Från: "DJB International" <djbinter(a)gmail.com> Datum: tors, dec 9, 2010 17:25 Rubrik: [Freeswitch-users] Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket) Till: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Are you running FS with -nocal? -djbinter On Thu, Dec 9, 2010 at 5:12 AM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: Ok - I trapped the problem with full event info. The up-time also reports a difference, in this case 32 seconds. As you can see there is no load on the server at all by this time, no calls etc. Hmm, this is really strange I must say. [2010-12-09 13:53:49] Received HEARTBEAT event - we're alive content-length == 629 content-type == text/event-plain core-uuid == 1550f368-b577-4304-bacd-2681d4067036 event-calling-file == switch_core.c event-calling-function == send_heartbeat event-calling-line-number == 65 event-date-gmt == Thu, 09 Dec 2010 12:53:53 GMT event-date-local == 2010-12-09 13:53:53 event-date-timestamp == 1291899233256303 event-info == System Ready event-name == HEARTBEAT freeswitch-hostname == 79w3sipt4 freeswitch-ipv4 == 10.66.195.14 freeswitch-ipv6 == ::1 idle-cpu == 98.437500 session-count == 0 session-per-sec == 30 session-since-startup == 277 up-time == 0 years, 7 days, 1 hour, 1 minute, 1 second, 680 milliseconds, 750 microseconds [2010-12-09 13:54:21] Received HEARTBEAT event - we're alive content-length == 631 content-type == text/event-plain core-uuid == 1550f368-b577-4304-bacd-2681d4067036 event-calling-file == switch_core.c event-calling-function == send_heartbeat event-calling-line-number == 65 event-date-gmt == Thu, 09 Dec 2010 12:54:25 GMT event-date-local == 2010-12-09 13:54:25 event-date-timestamp == 1291899265259303 event-info == System Ready event-name == HEARTBEAT freeswitch-hostname == 79w3sipt4 freeswitch-ipv4 == 10.66.195.14 freeswitch-ipv6 == ::1 idle-cpu == 97.656250 session-count == 0 session-per-sec == 30 session-since-startup == 277 up-time == 0 years, 7 days, 1 hour, 1 minute, 33 seconds, 683 milliseconds, 750 microseconds Mvh Peter Olsson -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Steven Ayre Skickat: den 9 december 2010 12:42 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] Reliability of HEARTBEAT events every 20 seconds? (in mod_event_socket) Also do you have any warning messages like this in the fs log? Task was executed late by %d seconds (That should show up if the delay is in the scheduler, if the task runs >1s late). -Steve On 9 December 2010 11:27, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: > Hello everyone, > > > > I have a question about the reliability of heartbeat events, it should be > sent from FreeSWITCH every 20 seconds - right? If you look below in my log > sample (from my own process), you can see that it comes 12:07:01, then 21, > then 41, but after this it suddenly takes 31 seconds for it to show up > (12:08:12). I've also done a Wireshark trace, and it shows the same thing - > so it is really delayed from FS. The load at the time was zero calls, and > the server itself was using maybe 5% CPU. After the delayed event it > continues to show up every 20 seconds. > > > > [2010-12-09 12:07:01] Received HEARTBEAT event - we're alive. > > [2010-12-09 12:07:21] Received HEARTBEAT event - we're alive. > > [2010-12-09 12:07:41] Received HEARTBEAT event - we're alive. > > [2010-12-09 12:08:12] Received HEARTBEAT event - we're alive. > > [2010-12-09 12:08:32] Received HEARTBEAT event - we're alive. > > [2010-12-09 12:08:52] Received HEARTBEAT event - we're alive. > > [2010-12-09 12:09:12] Received HEARTBEAT event - we're alive. > > > > Should the HEARTBEAT event be 100% reliable, or could something cause a > delay like this? This server is not on latest GIT, but it's not more then > one month old. I can debug this further myself, but I just want to know if > it always should be triggered every 20 seconds, or if these events for some > reasons might be delayed. > > > > By the way - no clock skew or anything is detected on the machine, so that's > not causing it. > > > > Thanks, > > > > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)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(a)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(a)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 !DSPAM:4d0102cf32763135019700!

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Refusing REFER requests
by mayamatakeshi 10 Dec '10

10 Dec '10

Is there any way to ask FS to refuse a REFER request? I mean, when a REFER is received, the first thing FS does is to disconnect the channel of the referrer. Then, it sends the other channel to the dialplan. But I want to prevent the unbridging of the call in some circumstances (meaning: some destinations for transfer are not valid). Obs: I don't want do disable support for REFER as I know it can be configured in the sofia profile. r, takeshi

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need help to install and use the Event socket library
by ha do 10 Dec '10

10 Dec '10

Hi all i follow the wiki page http://wiki.freeswitch.org/wiki/Event_Socket_Library the steps: cd ../libs/esl then vi python/Makefile and edit from LOCAL_CFLAGS=`python ./python-config --includes` LOCAL_LDFLAGS=`python ./python-config --ldflags` to LOCAL_CFLAGS=`python2.5 ./python-config --includes` LOCAL_LDFLAGS=`python2.5 ./python-config --ldflags` make pymod done then i switch to python mode do from freeswitch import * and get error: >>> from freeswitch import * Traceback (most recent call last): File "<stdin>", line 1, in <module> File "/usr/lib/python2.5/site-packages/freeswitch.py", line 7, in <module> import _freeswitch ImportError: No module named _freeswitch how do i install the esl for python properly thank you Ha`

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need help to install and use the Event socket library
by ha do 10 Dec '10

10 Dec '10

Hi all i follow the wiki page http://wiki.freeswitch.org/wiki/Event_Socket_Library the steps: cd ../libs/esl then vi python/Makefile and edit from LOCAL_CFLAGS=`python ./python-config --includes` LOCAL_LDFLAGS=`python ./python-config --ldflags` to LOCAL_CFLAGS=`python2.5 ./python-config --includes` LOCAL_LDFLAGS=`python2.5 ./python-config --ldflags` make pymod done then i switch to python mode do from freeswitch import * and get error: >>> from freeswitch import * Traceback (most recent call last): File "<stdin>", line 1, in <module> File "/usr/lib/python2.5/site-packages/freeswitch.py", line 7, in <module> import _freeswitch ImportError: No module named _freeswitch how do i install the esl for python properly thank you Ha`

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how bind_meta_app works?
by lakshmanan ganapathy 10 Dec '10

10 Dec '10

Hi all, I was experimenting the bind_mea_app and I have a doubt. There is a call from 1000 to FreeSwitch extension and it is connecting to event outbound socket. I've the following script in place. #!/usr/bin/perl use strict; use warnings; use Data::Dumper; use lib '/usr/src/freeswitch/libs/esl/perl/'; use IO::Socket::INET; use ESL; my $SOCK = new IO::Socket::INET ( LocalHost => 'localhost', LocalPort => '8447', Proto => 'tcp', Listen => 10, Reuse => 1 ); unless ($SOCK) { print("Could not create socket: $!"); exit(2); } while (1) { # Wait for any client through accept function in socket module. my $new_sock = $SOCK->accept(); next if (not defined($new_sock)); # Get socket host my $host = $new_sock->sockhost(); print("Got a client accepted from $host\n"); my $pid = fork(); if ($pid) { close($new_sock); next; } my $fd = fileno($new_sock); print("Newly forked child, pid: $$\n"); my $EslCon = new ESL::ESLconnection($fd); print "Connection created successfully\n"; my $info = $EslCon->getInfo(); $EslCon->setEventLock("true"); my $uuid = $info->getHeader("unique-id"); print Dumper $info->serialize(); $EslCon->execute("set","bind_meta_key=#"); $EslCon->execute("bind_meta_app","1 b o event::appli=testing"); my $api = $EslCon->execute("bridge","{ignore_early_media=true}freetdm/1/a/xxxxxxxxx"); } The script called to the number that was dialed. In that number I pressed #1, but the event was not sent, and I assume that the bridge application is executing in the A leg ( is it correct?? ) . So only after that bridge application gets completed the event gets triggered out. If I execute the application in the B leg, then it is working fine, since there is no application that is running at that time. Now I wanted to know if there is any way to run the application on A leg, I need that application to be executed even the bridge is not completed.

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