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Dialplan Issue
by Shelby Ramsey 02 Oct '09

02 Oct '09

Hello: I asked this on IRC yesterday and I think I confused everyone involved. So I apologize in advance here for reposting the question and if I wasted anyone's time. So here is the issue I'm having. I'm trying to use FS as a redirect server (specifically to serve up LNP queries via 302 redirects). But I'm having an issue where based on the string in the dialplan FS will respond with a 500 internal error message instead of a 300 redirect. The call flow should be this: -- remote party sends an Invite to my FS instance -- FS should respond with a 302 The following works as expected (FS will send a 302 when it receives an Invite): <action application="redirect" data="sip:${destination_number}@ ${network_addr};rn=${rn};npdi=yes"/> However if I do this (which is the way the response should look) FS will respond with a 500 internal server error: <action application="redirect" data="sip:${destination_number};rn=${rn};npdi=yes@${network_addr}"/> So the issue is the placement of the user params .... if they are before the @ FS will send a 500 internal server error ... if they are after the @ FS will send a 302. Unfortunately placing the user params after the @ doesn't quite conform to the way other devices expect to receive the 302 for this application. Any help would be greatly appreciated. Shelby PS ... hats off to the author of mod_memcache ... that is extremely useful!

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conference participant from behind NAT
by RobertT 02 Oct '09

02 Oct '09

I am a bit confused with what's going on in a following scenario. I have a public FS server with a public conference, that clients are connecting to with my softphone. All of this softphones have STUN option enabled and working, effectively resolving client's public IP address. They also have ICE enabled (but I guess it's not relevant here, since FS doesn't do ICE). Also, media trafic is secured with SRTP. The problem is when one client connects from port-restricted NAT into a conference he can hear sound for some time and he can be heard by other participants, but after awhile sound is gone and neither he hear anything nor he can be heard. Where is the problem? Is it NAT, closing RTP port after some silence period from client? I tried to start conference with waste flag, but without success eventually. The very same person can be contacted through this FS with direct call (being established in proxy_media mode) without any problems, but this is where ICE stuff starts doing its' "magic", I guess. Maybe I should try the same with SRTP disabled? Any help would be apreciated! Best regards, Robert.

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Minimum audio length for uuid_record
by Dan Le 02 Oct '09

02 Oct '09

We're running into a problem with the minimum file size when recording using uuid_record. It seems if the audio is too short it deletes the audio file. Is there a way to override that? We're running the FS 1.0.4 tag, on a windows xp 64bit machine. We start uuid_record from a lua script, recording at 8khz, if the recording is shorter than about 5s, the file gets blown away, but we may want to keep recordings that are 3-4s long. Thanks, Dan

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bgapi jobid to uuid
by Matt Riddell 01 Oct '09

01 Oct '09

Hi, I decided to go with a linked list for current channels and maintain that through state changes. So, basically it works like this: 1. Originate a call using bgapi (we get a jobid in response) 2. Receive an event with the jobid and a uuid 3. Lookup the linked list for the jobid, set the uuid 4. Receive hangup etc (with uuid) remove from linked list The problem is that sometimes I receive the hangup with the uuid before I'm told what the match between the jobid and the uuid are. Any ideas? -- Cheers, Matt Riddell Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

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REGISTER fails with 407 after minutes of success register
by Fernando Testa 01 Oct '09

01 Oct '09

Hi all, I have a FS that registers on an Ericsson pabx as gateway under sip_external. This gateway start registering on the Ericsson ok, but after a while, around 50mins, it fails with the logs below. If I hit *sofia profile external restart* on fs_cli then the gateway returns to register with success (that means, we get 200 OK from Ericsson). This happens with FS 1.0.4 release tarball, and trunk r15011. I found similar situations on these links, but not actually found a solution. Any help is very welcome. *OS* CentOS 5.3 x86_64 4Gb RAM log at http://pastebin.freeswitch.org/10517 conf/sip_profiles/external/ericsson.xml: <gateway name="ericsson_1064"> <param name="username" value="1064"/> <param name="realm" value="10.227.0.3"/> <param name="password" value="5080"/> <param name="register" value="true"/> <param name="register-proxy" value="10.227.0.3"/> <param name="expire-seconds" value="60"/> <param name="retry-seconds" value="10"/> </gateway>

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Listening to a connected call [barge in]
by Nagalenoj 01 Oct '09

01 Oct '09

In ES outbound, I need to do the following, * A calls 2000(FS ES outbound extension) * In the script, It'll answer the call, play some files and get the reply from A(as voice). * Simultaneously(when doing the above), the script has to call B. * When B attends the call, B has to listen to the live conversation between 2000 & A. How should I do.? I've tried this with async mode and by listening to the events. But I couldn't do it. Help me to do this.. Regards, Nagalenoj -- View this message in context: http://www.nabble.com/Listening-to-a-connected-call--barge-in--tp25696889p2… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Ringback when running G729 codec
by Hound Dog 30 Sep '09

30 Sep '09

Hi, very happy with freeswitch as a PBX/softswitch/SBC system its working solidly for a few weeks now - just great I have a question regarding ringback tones - custom or regular - I cant get freeswitch to send ringback using G729 I used the following settings ( it will just play one of the IVR prompts as ringback (filename ivr-to_repeat_these_options) - I took it from the G729 encoded files package , it has PCMA , G729 G723 extensions ) <extension name="inbound_routing"> <condition field="destination_number" expression="^4420885767(0\d)$"> <action application="set" data="ringback=/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options"/> <action application="set" data="instant_ringback=true"/> <action application="bridge" data="user/10$1"/> </condition> </extension> when I call with G711 enabled , it plays the file no problems - see log 2009-09-25 11:29:58.641361 [DEBUG] sofia.c:3289 Channel sofia/external/442078562107(a)80.80.80.80 entering state [early][183] 2009-09-25 11:29:58.641361 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562107(a)80.80.80.80 [BREAK] 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1718 Raw Codec Activation Success L16@8000hz 1 channel 20ms 2009-09-25 11:29:58.641361 [DEBUG] switch_ivr_originate.c:1745 Play Ringback File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 2009-09-25 11:29:58.641361 [INFO] mod_native_file.c:82 Opening File [/usr/local/freeswitch/sounds/en/us/callie/raw_files/ivr/ivr-to_repeat_these_options.PCMA] 8000hz 2009-09-25 11:29:58.889369 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1006@82.80.131.233:40505 entering state [proceeding][180] when I call with G729 only - I get silence , and freeswitch only send the comfort noise packet and no RTP , see log 2009-09-25 11:28:57.437537 [DEBUG] sofia.c:3289 Channel sofia/external/442078562107(a)80.80.80.80 entering state [early][183] 2009-09-25 11:28:57.437537 [DEBUG] switch_core_session.c:630 Send signal sofia/external/442078562107(a)80.80.80.80 [BREAK] 2009-09-25 11:28:57.685536 [DEBUG] sofia.c:3289 Channel sofia/internal/sip:1006@82.80.131.233:40505 entering state [proceeding][180] 2009-09-25 11:28:57.685536 [NOTICE] sofia.c:3353 Ring-Ready sofia/internal/sip:1006@82.80.131.233:40505! mod_native_file works well for me when used in applications and plays G729 files no problem any ideas why is that happening , any suggestions on how to resolve ? thanks Ori

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Re: [Freeswitch-users] Need to know
by Ahmed Munir 30 Sep '09

30 Sep '09

Hi, Thanks for reply. Actually I'm currently running asterisk and for billing I'm using MySQL database. In the database i've listed various destination against their different rates. When call is made a script which queries MySQL for billing. My question is, how can I do this in FS using mod nibbles(note: I'm running asterisk in realtime)? And my other question is, will MySQL authenticates each time when call is made on FS? Kindly advise me. > ---------- Forwarded message ---------- > From: Ahmed Munir <ahmedmunir007(a)gmail.com> > To: FreeSwitch <freeswitch-users(a)lists.freeswitch.org> > Date: Wed, 30 Sep 2009 10:28:41 +0600 > Subject: [Freeswitch-users] Need to know > Hi, > > I want to use mod_nibble in freeswitch for billing purpose. I've got some > questions, as I'm listing down below; > > 1- How I create MySQL connection with FS? > 2- Is it possible to query the database when billing occurs? > 3- Will database credentials require each time when query to database? > 4- I'm currently running MySQL database in production for billing > purposes.Will it be okay to use my current database for mod_nibble? > > Kindly advise me soon and also please provide me the links as well. > > -- > Regards, > > Ahmed Munir > > > > > ---------- Forwarded message ---------- > From: William Suffill <william.suffill(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Wed, 30 Sep 2009 00:43:37 -0400 > Subject: Re: [Freeswitch-users] Need to know > mod_nibble uses ODBC. Via ODBC u can connect to mysql. > > http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core > > mod_nibble has it's own table design. > http://wiki.freeswitch.org/wiki/Mod_nibblebill > > It may be possible to tweak it to use your current tables but I'd > suggest getting the > basics working based on the documentation on the wiki and build from there. > > -- W > > > > -- Regards, Ahmed Munir

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Attention: behavior change with r15025
by Tamas 30 Sep '09

30 Sep '09

Hello, At r15025 the ringback tone handling has changed. The old method: When ringback was set, early media was ignored, meaning that if you got 183 with SDP, ringback was generated locally. The new way: When ringback is set, the ringback tone is generated only when there is no early media (thus 180 w/o SDP or 183 w/o SDP). When early media is presented, it will be used as ringing indication. To get the same behaviour as before r15025, you have to set the following: ignore_early_media=true ringback Some wiki pages were changed to report the new behavior. Many thanks for guys for fix/feature, especially for anthm and bkw. Kind regards, Tamas ps.: issue was related with FSCORE-454.

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Multiple Subnets
by Andrey Nepomnyaschih 30 Sep '09

30 Sep '09

Hello, I have some problems running a FreeSWITCH with endpoints located in different subnets. For example a FS is listening at 192.168.50.14/32 and endpoints from the same (192.168.50.0/24) subnet work as expected. But when I try to receive a media from an endpoint located at different subnet, let's say 192.168.60.0/24 the RTP stream cannot be bridged. FS sends an INVITE message asking endpoint to send media to external interface. Can anyone give me a hint where to look on how to overcome such behavior? Kind Regards, Andrey Nepomnyaschih

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