December 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Redirecting a call from one FS to another FS?
by Dennis 19 Dec '08

19 Dec '08

i would like to know, what the best way is, to redirect an incoming call from one fs (fs1) to another fs (fs2). we use 3 freeswitch servers and the carrier passes calls to the three fs servers randomly. if on fs server is not offline, the carrier sends the call to the next fs. this is generally good, but for conferencing it not so good. i am using socket outbound and need to do this for conferencing. let's say, we have a conference going on on fs1. another person wants to enter this conference, but the call is passed to fs2. on fs2 we see, that the caller wants to enter the conference going on on fs1. now we have to redirect the call from fs2 to fs1. is this done with "redirect" and some according settings/params or are there other ways to do this? we would like to do this without our carrier doing something, to be a little more independant. thanks dennis

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Ending a bridged call with a touchtone
by Marc Orenberg 19 Dec '08

19 Dec '08

Thanks for the response Brian. I don't understand what bind_meta does, or how it can help me. Is it something I can use from my Python script? I searched for a description of it, but I was unable to find one. Could you please point me towards some documentation, or maybe quickly explain it? Thanks. >Date: Tue, 16 Dec 2008 11:30:46 -0600 >From: Brian West <brian(a)freeswitch.org> >Subject: Re: [Freeswitch-users] Ending a bridged call with a touchtone >To: freeswitch-users(a)lists.freeswitch.org >Message-ID: <F9E12702-5CA0-45A8-B408-3F2162F94A8E(a)freeswitch.org> >Content-Type: text/plain; charset="us-ascii" > >Try bind_meta, examples are in the default dialplan. > >/b > >On Dec 16, 2008, at 10:24 PM, Marc Orenberg wrote: > >> Hello. I'm trying to allow the A-leg of a bridged call to be able >> to press a touchtone to end the call. >> In my Python script, I set-up a DTMF callback function using >> setInputCallback, but it doesn't seem to have any effect during >> bridged calls. Is there another way to do this? Thanks.

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Java example
by kriko 19 Dec '08

19 Dec '08

I made a simple java example, following this guide http://wiki.freeswitch.org/wiki/Java so when someone calls it should print something in console. I've also modified dialplan/public.xml, what I want is to intercept calls from jabber and process them: http://pastebin.com/m35de11d9 But there is nothing printed into console, path to jar is ok, also class name is ok. Java module is being loaded on startup. What could be wrong? -- kriko

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Re: [Freeswitch-users] Core Dump
by Michael S Collins 19 Dec '08

19 Dec '08

Check out this page: wiki.freeswitch.org/wiki/Debugging_Freeswitch -MC Sent from my iPhone On Dec 18, 2008, at 6:38 AM, "peder(a)networkoblivion.com" <peder(a)networkoblivion.com > wrote: > What is the process for capturing and submitting a core dump? > > I am messing around with the Cisco 79x1 phones and tcp and multiple > reg. > I have a 7961 using tcp and a 7960 using udp both reg'd with the same > number and both showing up as registered. If I call out from the > phone > using tcp, it works. If I call out from the phone using udp, I get a > core dump. If I call in, it calls both phones and I am assuming the > call to the phone using udp causes a core dump as well. These are the > only two phones on the system and I am running version 10851 from > yesterday. If I only have the udp phone registered and the tcp > phone is > off, it works fine. It is only when I have a mix of a udp reg and a > tcp > reg on the same number that I appear to get a core dump. > > > Peder > > _______________________________________________ > 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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Pennytel Gateway Registration problem
by Scott Ellis 18 Dec '08

18 Dec '08

I have a standard install, and I am trying to get a Pennytel gateway to register. After looking at Wireshark traces of x-lite registering and FreeSwitch registering, FreeSwitch is not sending any authentication information with the registration request. I am obviously missing something here! I understand for incoming calls you don't want authentication, but for outgoing it is obviously required. Is there a flag somewhere that I am supposed to set? The file was taken from the wiki page, and looks like it was previously tested when using the obsolete outbound directory structure. The following file is in the conf/sip_profiles/external directory. <include> <gateway name="PennyTel"> <param name="username" value="888917xxxx"/> <param name="password" value="xxxxxxx"/> <param name="realm" value="sip.pennytel.com"/> <param name="proxy" value="sip.pennytel.com"/> <param name="register" value="true"/> <param name="expire-seconds" value="60"/> </gateway> </include> Thanks. Scott

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Crackling noise when bypassing media between endpoints.
by mszlazak＠aol.com 18 Dec '08

18 Dec '08

When using bypass_media (aka. no_media) mode between an X-lite softphone and Prophacy ASR, I get intermittent "crackiling" background noise with the audio that I'm hearing. How do I get rid of this?

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mod_shout and mp3 formats
by Peter P GMX 18 Dec '08

18 Dec '08

I try to play mp3 I generated through Cepstral TTs and which I encoded via lame. However they won't play, so my question is: Which mp3 formats are supported? I generate the wav files by the following /opt/swift/bin/swift -n Katrin -p audio/channels=1,cst/f0_shift=.8,speech/rate=120,audio/sampling-rate=8000,audio/deadair=2 -o $wavefile $text Then I convert to mp3 by the following variations: lame 46.wav 46.mp3 lame -s 32 46.wav 46.mp3 lame --preset 128 46.wav 46.mp3 lame --resample 44.1 --preset 128 46.wav 46.mp3 lame --resample 32 --preset 128 46.wav 46.mp3 lame --resample 44.1 46.wav 46.mp3 lame --resample 44.1 -m s --preset 128 46.wav 46.mp3 lame --resample 44.1 -m s 46.wav 46.mp3 lame --resample 44.1 -m s -b 128 46.wav 46.mp3 lame --resample 44.1 -m s -B 24 46.wav 46.mp3 lame --preset voice -v -B 64 -a 46.wav 46.mp3 None of them worked with the playback application (shout://localhost/tts/46.mp3). The sound files had a length of between 2 and 5 sec. 2 Times during various tries they played at least partially. But at the next try they didn't play again. However I have a prerecorded sound file (44.1KHz, 128 kBits stereo music) which always plays well. The console shows me that all files are successfully played and I get a channel_ececute and a channel_ececute_complete after some seconds during event_socket. But I don't hear any sound. All above samples however played well with Totem on Ubuntu. The wiki tells me that almost any mp3 format should play. What am I doing wrong here? Another question: Should normal wav files play as well? Also with wav I cannot hear any sound. Best regards Peter

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Gtalk to sip problems when reconfiguring from scratch
by kriko 18 Dec '08

18 Dec '08

I recently purged all freeswitch config and restarted configuring from scratch. Using defaults, I modified public.xml dialplan config (added line 16 - 28): http://pastebin.com/m5ece6e6f and added a new config under jingle_profiles: http://pastebin.com/d6e983b99 I register with phonelite or twinkle as user 1000 (it says successfull reg.), but when I make a call from gtalk I hear that user is not available, none of client rings. When registering sofia prints: 2008-12-18 15:53:32 [INFO] sofia_presence.c:475 actual_sofia_presence_event_handler() internal START_PRESENCE_PROBE_SQL 2008-12-18 15:53:32 [NOTICE] sofia_presence.c:793 sofia_presence_resub_callback() internal PRESENCE_PROBE 1000(a)10.99.8.221 2008-12-18 15:53:32 [INFO] sofia_presence.c:484 actual_sofia_presence_event_handler() internal END_PRESENCE_PROBE_SQL 2008-12-18 15:53:32 [INFO] sofia_presence.c:547 actual_sofia_presence_event_handler() IN START_PRESENCE_SQL (internal) 2008-12-18 15:53:32 [INFO] sofia_presence.c:563 actual_sofia_presence_event_handler() IN END_PRESENCE_SQL (internal) 2008-12-18 15:53:32 [WARNING] sofia_presence.c:517 actual_sofia_presence_event_handler() external is passive, skipping 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510 actual_sofia_presence_event_handler() 10.99.8.221 is an alias, skipping 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510 actual_sofia_presence_event_handler() default is an alias, skipping 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510 actual_sofia_presence_event_handler() nat is an alias, skipping 2008-12-18 15:53:32 [WARNING] sofia_presence.c:510 actual_sofia_presence_event_handler() outbound is an alias, skipping and when fetching registrations: 2008-12-18 15:53:50 [ERR] sofia_reg.c:1120 sofia_reg_handle_sip_i_register() NO CONTACT! Does this means that registration failed? Why it doesn't call anymore the targeted user (1000)? Here is also an extract from console log while calling, it caught my attention: 2008-12-18 15:54:41 [WARNING] mod_dptools.c:2047 user_outgoing_channel() Can't find user [1000@] 2008-12-18 15:54:41 [ERR] switch_ivr_originate.c:1110 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2008-12-18 15:54:41 [DEBUG] switch_ivr_originate.c:1689 switch_ivr_originate() Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] -- kriko

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Core Dump
by peder＠networkoblivion.com 18 Dec '08

18 Dec '08

What is the process for capturing and submitting a core dump? I am messing around with the Cisco 79x1 phones and tcp and multiple reg. I have a 7961 using tcp and a 7960 using udp both reg'd with the same number and both showing up as registered. If I call out from the phone using tcp, it works. If I call out from the phone using udp, I get a core dump. If I call in, it calls both phones and I am assuming the call to the phone using udp causes a core dump as well. These are the only two phones on the system and I am running version 10851 from yesterday. If I only have the udp phone registered and the tcp phone is off, it works fine. It is only when I have a mix of a udp reg and a tcp reg on the same number that I appear to get a core dump. Peder

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Re: [Freeswitch-users] Call sip phones from gtalk / jabber
by Kirk Bateman 18 Dec '08

18 Dec '08

Brian, That wasn't exactly what I meant :) I have had Freeswitch connecting to GTalk directly as a client and that was where I was getting the issues with sending anything to other domains. I haven't actually tried the server profile with my own ejabberd server. What I was planning to do was have a separate ejabberd bot which is the control interface which then sends bits to the freeswitch connected "bot" / user so I can sanitise anything before it gets to freeswitch, as I have found the freeswitch jabber stuff to segfault every now and then (that said I haven't updated in about a month or so). Just haven't had enough time to actually look into what was causing the segfaults. I did get the gtalk to sip bits working though, had a few codec issues but gtalk to freeswitch worked fine, unfortunately my slightly more complicated gtalk to freeswitch to asterisk to SPA941 has some codec bits to work out, I think its to do with asterisk PCMU being different ? I know I had something similar when trying to use the same setup with gizmo (which I fixed by forcing G729 from SPA941 -> asterisk -> freeswitch. Cheers Kirk > Date: Wed, 17 Dec 2008 11:21:32 -0600 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] Call sip phones from gtalk / jabber > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <876638A9-FBB0-4079-A76F-814E6953D395(a)freeswitch.org> > Content-Type: text/plain; charset="us-ascii" > > FreeSWITCH already logs into your jabber server as a component if you > cant communicate with other domains then your jabber server is not > configured correctly. > > /b > > On Dec 17, 2008, at 11:15 AM, Kirk Bateman wrote: > > > Kriko, > > > > I have been looking at the same sort of thing, but I'm planning to > > implement an ejabberd bot component (so I can hopefully use the new > > mod_erlang_event freeswitch interface). > > > > It seems to me that bits of the current dingaling / jingle interface > > are having problems, like not liking sending messages to other > > domains, its generally working if they are all .gmail.com users but > > when you have some ongooglemail.com etc it starts breaking, and > > doesn't use the whole JID in the from attribute for sending messages. > > > > When I get a chance I'll try and narrow down the problem. > > > > Cheers > > > > Kirk > >

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