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no Ringing-Signal on incomming calls via mod_freetdm/sangoma_isdn
by Helmut Kuper 02 Nov '11

02 Nov '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I switched from old openzap PRI-Stack to freetdm with sangoma_isdn because of some odd problems with our EWSD switch. Setup was quite easy. Only one thing is strange. When I call into FS from pstn to a sip phone I get no ringing resp ALERTING signal to my pstn phone. Executing ring_ready in dialplan just befor bridging works. Unfortuantely it shows the same behaviour as using "instant_ringback=true", e.g. First ringing then busy ... I had that problem also using old openzap. So today I looked around in FS code and found that when mod_sofia sends a SWITCH_MESSAGE_INDICATE_RINGING on receiving a sip 180 reply back to the caller's channel it works. Is there any reason why sofia doesn't do this by default? Oh, I use FS git-1d3417a 2011-06-07 17-35-49 -0400. best regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk6xpEoACgkQ4tZeNddg3dxGjwCePxDkj4fVdprToKPov5+OeM/3 YakAnitvJZod29hSW4J9T/Sn2Vuob0rU =4KAd -----END PGP SIGNATURE-----

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ekiga, celt codec, Freeswitch and audio problems
by Helmut Kuper 02 Nov '11

02 Nov '11

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I tried to setup a phone call using celt codec from ekiga to freeswitch to ekiga, where fs doesn't bypass media. ekiga: 3.2.7 (codec list is CELT@48000, CELT@32000, G722, PCMA) Freeswitch: git-a14b20a 2011-09-24 09-58-04 -0500 celt: 0.7.1 (Unfortunately this is what ekiga is using) When FS bypasses the media everything works fine. When ekiga is calling into FS and FS plays a mp3 file to it, it sounds fantastic on ekiga side. When FS doesn't bypasses media the spoken words are distorted. When I use G722 with ekiga and FS in between, there is clear audio. I played around with rtp timer and rewrite_timestamps, but no success. Is it because of celt 0.7.1? Any ideas where to look further? regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.10 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk6xdFwACgkQ4tZeNddg3dwVfQCdGIX+BSksJp9gfW5C5QWqrS/i zBAAoLm5i8n7wGaEkIFkOKolUPgM3zYE =8WAc -----END PGP SIGNATURE-----

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double dtmf
by Yuriy Nasida 02 Nov '11

02 Nov '11

Hello community, I have some issue with delivery of dtmf.The call flow is:one sip provider - FS - second sip provider - endpoint which expects correct dtmf On FS I have simple custom lua application which implement logic of calling cards system. I have not issues when I call to lua application. I can enter destination number (by using dtmf apparently) and all works fine. But when I try to send dtmf second time for endpoint which expects correct dtmf I have issue with double dtmf. It is noted that when I call from sip device I have not ANY issues. But I have issue when I call from any cell phone or landline phone. Any ideas are welcomed. Thanks.

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LCR and Continue On Fail
by Antonio Teixeira 02 Nov '11

02 Nov '11

Hello Guys. Logs & Dialplan http://pastebin.freeswitch.org/17613 I'm currently trying to connect LCR with the Fall back provided with Continue On Fail. I could be misunderstading the use of continue on fail i read several posts on the mailing and in theory this should work :) Anyway my objective is that If failure _cause == UNALLOCATED_NUMBER continue using the lcr reach else: hangup the call. So this is me or could be a bug ?? Regards & Thanks for the help :) Antonio Teixeira

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Public IP in Contact header
by Hynek Cihlar 02 Nov '11

02 Nov '11

Can anybody please point me to settings or a freeswitch state that makes sofia put public IP into the Contact header? Currently I have SIP clients behind NAT registering to freeswitch. Freeswitch registers these clients with their respective private IPs. Bellow is the profile status. ================================================================================================= Name internal Domain Name N/A Auto-NAT false DBName sofia_reg_internal Pres Hosts 178.238.X.Y,178.238.X.Y Dialplan XML Context default Challenge Realm auto_from RTP-IP 178.238.X.Y Ext-RTP-IP 178.238.X.Y SIP-IP 178.238.X.Y Ext-SIP-IP 178.238.X.Y URL sip:mod_sofia@178.238.X.Y:5060 BIND-URL sip:mod_sofia@178.238.X.Y:5060;maddr=178.238.X.Y HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU,GSM,G7221@32000h,G7221@16000h,G722 CODECS OUT PCMA,PCMU,GSM,G7221@32000h,G7221@16000h,G722 TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: 3454998XYZ@192_168_59_102 User: 1@ribtjuhd Contact: "383372XYZ" <sip:1@192.168.59.111:5060> Agent: A580 IP/022270000000 Status: Registered(UDP)(unknown) EXP(2011-11-02 13:30:43) EXPSECS(100) Host: eva IP: 81.90.X.Y Port: 22055 Auth-User: 1 Auth-Realm: ribtjuhd MWI-Account: 1@ribtjuhd Thanks! Hynek

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FreeSWITCH Conference Call Today - New Dialplan Feature, SIP 101
by Michael Collins 02 Nov '11

02 Nov '11

Hello all! In our conference call today we are going to discuss a cool new dialplan feature (better logical OR processing) and then Ken Rice and I will be discussing some basic SIP knowledge to help get everyone up to speed when helping newbies. Talk to you all soon! -Michael

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bridging calls from different sofia profiles
by Julien Chavanton 01 Nov '11

01 Nov '11

Hi, I have audio problems when bridging calls from different sofia profiles. I use 2 profiles in order to connect softphone through a VPN to avoid NAT problem One profile is sending/listening on a private IP (for softphone) and the other profile is using a public IP (for external PSTN gateways) I have tested with X-lite and Ekiga and I face different one way audio with both: With X-lite the audio received from the private IP is not forwarded While with Ekiga the audio from the public IP profile is not forwarded I have traced SIP and RTP and the signaling looks good. RTP is received by FS on the correct socket but not forwarded. I understand I do not provide enough information to diagnostic, but maybe I could get input on how to diagnostic RTP socket etc. in this scenario ?

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Log level of log file by fs_cli
by fieldpeak 01 Nov '11

01 Nov '11

Hi friends, i know we can set the loglevel inside switch.conf.xml to change the log level of log files, however, i would like to change the log level of log files thru fs_cli on the fly(no need restart FS), is there any commands to support it, thanks. -- Regards, Charles

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Using Presence with Valet Parking
by Spencer Thomason 31 Oct '11

31 Oct '11

Hello, I noticed there have been some recent improvements to mod_valet_parking regarding presence. I was wondering if anyone had used this already and could shed some light on how to subscribe to the status of a particular slot. I'm testing this with a Cisco SPA-509G and can't seem to get it working. My dialplan looks like this: <extension name="parking_slots"> <condition field="destination_number" expression="^(7[1-4])$" break="on-false"> <action application="answer"/> <action application="valet_park" data="parking_slots $1"/> </condition> </extension> The phone is sending a SUBSCRIBE to park+71@<DOMAIN> and gets a 202 Accepted response. I do have presence configured and working correctly for user extensions. When a call is transferred to 71, the parking works as expected but a NOTIFY is never sent to the phone. What am I missing? Thanks for any help! Spencer

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Re: [Freeswitch-users] RTP Media Port = 0 Bug
by chrisbware＠interfree.it 31 Oct '11

31 Oct '11

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