- FreeSWITCH-users - FreeSWITCH Mailing Lists

Downloading Free Switch
by Shaik basha 22 Mar '10

22 Mar '10

Dear All, Good Afternoon every one in this forum. I am new to FS. I tried to understand about FS, but could not understand completely. Hence would any one help me from where I can download FS and what FS can do exactly. Your early response in this regard would be very much appreciated. Thanking you in advance. Regards, shaikh.bashaa

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Open source softphone that can handle g729
by Fraser Redmond 21 Mar '10

21 Mar '10

We're looking for an open source softphones that can handle g729 that we can extend with extra functionality (connecting to other applications, and things on the intranet.) Does anyone have any they'd suggest? Brian West suggested FSComm with the qualifier "it still needs more work" - has anyone used it much (especially in windows), or know what it's development status is? (Looking in the source browser jmesquita spent a lot of time on it in January, but not much since.) Cheers, Fraser

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Re: [Freeswitch-users] rtp timeout yet
by Madovsky 21 Mar '10

21 Mar '10

Apparently there s a timeout everytime I register a phone on a public IP. for phone behind a firewall it works well. ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, March 21, 2010 3:09 PM Subject: rtp timeout yet last trunk, remove current conf folder and reinstalled the default one after make install. I stopped iptables, connect 1000 to 1001 on port 5060, but after rtp-timeout it closes yet. sip trace after the rtp timeout SIP/2.0 481 Call Does Not Exist I start to be very frustrated....

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Freeswitch SPA3000 HUP Not Sent
by Ben Langfeld 21 Mar '10

21 Mar '10

Hey, I have a small 7 seat mostly softphone based PBX installation in place using freeswitch and a couple of SPA3000s for PSTN termination. Currently, aside from PSTN HUP detection, outbound calls are fine setup to directly dial the SPAs. When the internal VoIP side hangs up, freeswitch sends NORMAL_CLEARING to the ATA and the ATA drops the call. Lovely. Unfortunately inbound calls aren't so successful. Calls are sent using the SPA dialplan to an internal freeswitch extension, and the calls connect fine. This time, when the VoIP side hangs up, no NORMAL_CLEARING is sent to the ATA. Is there a reason for this? Can anyone give me any idea how I can get freeswitch to instruct the SPA to drop the line? Regards, Ben Langfeld Wave > Email

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Paging
by Sean Holt 21 Mar '10

21 Mar '10

Hello Dev team, Freeswitch rocks!! I have a question about the paging application. I¹ve followed the instructions on the wiki on how to set it up, but when I do a sip trace I¹m noticing a Proxy authentication message when try to connect to the paging extension. Every time I dial the extension it hungs up. I¹ve checked the acl xml file and my subnet is in the access list. Any thoughts on how to correct the authentication issue, if that¹s issue. Thanks Sean

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not registered external softphone's with internal profile
by Vladimir Elizarov 21 Mar '10

21 Mar '10

Hello, I'm trying to register on the fs external softphone (with public ip). Copied profine internal to internal5090 (http://pastebin.freeswitch.org/12506) When you register get 405 Method Not Allowed or 481 Subscription Does Not Exist (http://pastebin.freeswitch.org/12507 and http://pastebin.freeswitch.org/12508) Depending on what softphone use. sofia_contact shows that the username is not registered (error/user_not_registered), although it is in sofia status internal5090 (http://pastebin.freeswitch.org/12509) What causes the appearance of 405, 481 and the absence of registration? -- Best regards, Vladimir Elizarov

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openzap no audio?
by Tom Christensen 21 Mar '10

21 Mar '10

I've got freeswitch installed with a digium te110p hooked up to a PRI. calls through the system work fine SIP->SIP, calling and receiving calls over the PRI works, (IE the signalling part, a call can be made and incoming calls come in over the PRI and I can route them based on DID), however, there is no audio to either party. I've checked, and there is an RTP stream running while the call is up, there is no reason it would be blocked (IE, SIP->SIP calls use the same RTP port range, and they work fine). I saw a few posts that talked about no audio for a few seconds after a call is connected, or that calls would connect and then drop after a few seconds. This is different, I can leave the call up for many minutes, it doesn't drop, and audio never starts coming through. I'm running openzap natively, no libpri. I am using the dahdi 2.2.1, tried the latest release of zaptel first, freeswitch wouldn't even load with that. I'm going to try libpri now I think as I've exhausted all the other options I can think of. Any input/ideas would be greatly appreciated. -Tom

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I need help about my gateway
by Александр Эдуардович 21 Mar '10

21 Mar '10

Hello! I tried to search information about my problem at the internet and local sites/forums, but I didn't find solution. My provider need in CONTACT field: "7455(a)providerserver.com" (7455 - is login on provider server), but FS send: gw+gwname(a)mydomain.com<gw%2Bgwname(a)mydomain.com> ". If I use option "<param name="extension-in-contact" value="true"/>" in my gateway config - FS send this: "7455(a)mydomain.com". As a result, my external calls didn't send over this provider: "2010-03-20 17:56:30.852222 [NOTICE] switch_channel.c:669 New Channel sofia/external/7451 [0ab674ec-bc8c-48e0-9235- eb395dfe02b0] 2010-03-20 17:56:30.915402 [NOTICE] sofia.c:4788 Hangup sofia/external/7451 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2010-03-20 17:56:30.918438 [NOTICE] switch_core_session.c:1179 Session 28 (sofia/external/7451) Ended 2010-03-20 17:56:30.918438 [NOTICE] switch_core_session.c:1181 Close Channel sofia/external/7451 [CS_DESTROY] 2010-03-20 17:56:30.919604 [INFO] mod_dptools.c:2353 Originate Failed. Cause: NO_ROUTE_DESTINATION 2010-03-20 17:56:30.919604 [NOTICE] mod_dptools.c:2416 Hangup sofia/internal/1000(a)mydomain.com [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2010-03-20 17:56:30.942906 [NOTICE] switch_core_session.c:1179 Session 27 (sofia/internal/1000(a)mydomain.com) Ended 2010-03-20 17:56:30.942906 [NOTICE] switch_core_session.c:1181 Close Channel sofia/internal/1000(a)mydomain.com [CS_DESTROY]" My gateway config: "<include> <gateway name="gwname"> <param name="username" value="7455"/> <param name="realm" value="providerserver.com"/> <param name="from-user" value="7455"/> <param name="from-domain" value="providerserver.com"/> <param name="password" value="pass"/> <param name="extension" value="7455"/> <param name="proxy" value="providerserver.com"/> <param name="register-proxy" value="providerserver.com"/> <param name="expire-seconds" value="160"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="retry-seconds" value="30"/> <param name="extension-in-contact" value="true"/> <param name="caller-id-in-from" value="true"/> <param name="contact-params" value="tport=tcp"/> <param name="ping" value="25"/> </gateway> </include> This gateway status in "sofia status": REGED. Sorry for my bad english ))) Best regards, Alexander

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Re: [Freeswitch-users] mod_voicemail modifications
by Anthony Minessale 20 Mar '10

20 Mar '10

I think I have said at least once. If you have that email delete problem Update to trunc and Issue make vm-sync You have an out-of-date vm template in your config. Once you have the correct template it will not even offer the email option when you don't have email. For the other reqs file a bounty or feature req jira for the desired options. On Mar 19, 2010 5:13 PM, "Robert Hadley" <robert.hadley(a)teotech.com> wrote: Hi Tim, I wrote up a Jira bug for the email key always deleting voicemails. Andrew T. responded with a patch that is supposed to fix half of the problem such that voicemails are not deleted when emailed. However he didn't include using the vm-keep-local-after-email setting in the user profile. I haven't had a chance to try this patch yet. See http://jira.freeswitch.org/browse/MODAPP-403 I would be interested in any voicemail fixes being made too. Regards, Robert -----Original Message----- From: Tim St. Pierre [mailto:fs-list@communicatefreely.net] Sent: Fri...

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dahdi problems
by Brian May 20 Mar '10

20 Mar '10

Hello, Just noticed zaptel was renamed dahdi. What am I doing wrong? 2010-03-17 19:11:32.505233 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/modules.conf. 2010-03-17 19:11:32.506233 [NOTICE] zap_io.c:3048 Modules configured: 1 2010-03-17 19:11:32.506233 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/openzap.conf. 2010-03-17 19:11:32.506233 [DEBUG] zap_io.c:2578 found config for span 2010-03-17 19:11:32.506233 [NOTICE] ozmod_zt.c:1182 Using DAHDI control device 2010-03-17 19:11:32.507233 [INFO] zap_io.c:2849 Loading IO from /opt/freeswitch/mod/ozmod_zt.so [zt] 2010-03-17 19:11:32.507233 [DEBUG] zap_config.c:56 Configuration file is /opt/freeswitch/conf/zt.conf. 2010-03-17 19:11:32.507233 [INFO] ozmod_zt.c:556 Setting rxgain val to 0.000000 2010-03-17 19:11:32.507233 [INFO] ozmod_zt.c:565 Setting txgain val to 0.000000 2010-03-17 19:11:32.508233 [INFO] zap_io.c:2595 auto-loaded 'zt' 2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2616 created span 1 (FXS1) of type zt 2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2629 span 1 [name]=[Brian May] 2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2629 span 1 [number]=[1002] 2010-03-17 19:11:32.508233 [DEBUG] zap_io.c:2629 span 1 [fxs-channel]=[1] 2010-03-17 19:11:32.509233 [DEBUG] zap_io.c:2671 setting trunk type to 'FXS' start(KEWL) 2010-03-17 19:11:32.509233 [ERR] ozmod_zt.c:269 failure configuring device /dev/dahdi/channel chan 1 fd 64 (No such device or address) 2010-03-17 19:11:32.509233 [ERR] zap_io.c:1781 I/O backend does not support command 32! 2010-03-17 19:11:32.509233 [ERR] zap_io.c:1781 I/O backend does not support command 34! 2010-03-17 19:11:32.509233 [DEBUG] zap_io.c:2578 found config for span 2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2616 created span 2 (FXS2) of type zt 2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2629 span 2 [name]=[Brian May] 2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2629 span 2 [number]=[1002] 2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2629 span 2 [fxs-channel]=[2] 2010-03-17 19:11:32.510233 [DEBUG] zap_io.c:2671 setting trunk type to 'FXS' start(KEWL) 2010-03-17 19:11:32.511233 [ERR] ozmod_zt.c:269 failure configuring device /dev/dahdi/channel chan 2 fd 64 (No such device or address) 2010-03-17 19:11:32.511233 [ERR] zap_io.c:1781 I/O backend does not support command 32! 2010-03-17 19:11:32.511233 [ERR] zap_io.c:1781 I/O backend does not support command 34! 2010-03-17 19:11:32.511233 [DEBUG] zap_io.c:2578 found config for span 2010-03-17 19:11:32.511233 [DEBUG] zap_io.c:2616 created span 3 (FXO1) of type zt 2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2629 span 3 [name]=[OpenZAP] 2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2629 span 3 [number]=[0397551926] 2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2629 span 3 [fxo-channel]=[3] 2010-03-17 19:11:32.512233 [DEBUG] zap_io.c:2658 setting trunk type to 'FXO' start(KEWL) 2010-03-17 19:11:32.512233 [ERR] ozmod_zt.c:269 failure configuring device /dev/dahdi/channel chan 3 fd 64 (No such device or address) 2010-03-17 19:11:32.512233 [ERR] zap_io.c:1781 I/O backend does not support command 32! 2010-03-17 19:11:32.513233 [ERR] zap_io.c:1781 I/O backend does not support command 34! 2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2578 found config for span 2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2616 created span 4 (FXO2) of type zt 2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2629 span 4 [name]=[OpenZAP] 2010-03-17 19:11:32.513233 [DEBUG] zap_io.c:2629 span 4 [number]=[0397552345] 2010-03-17 19:11:32.514233 [DEBUG] zap_io.c:2629 span 4 [fxo-channel]=[4] 2010-03-17 19:11:32.514233 [DEBUG] zap_io.c:2658 setting trunk type to 'FXO' start(KEWL) 2010-03-17 19:11:32.514233 [ERR] ozmod_zt.c:269 failure configuring device /dev/dahdi/channel chan 4 fd 64 (No such device or address) 2010-03-17 19:11:32.514233 [ERR] zap_io.c:1781 I/O backend does not support command 32! 2010-03-17 19:11:32.514233 [ERR] zap_io.c:1781 I/O backend does not support command 34! 2010-03-17 19:11:32.515233 [INFO] zap_io.c:2773 Configured 0 channel(s) 2010-03-17 19:11:32.516233 [ERR] zap_io.c:3056 No modules configured! 2010-03-17 19:11:32.516233 [ERR] mod_openzap.c:3171 Error loading OpenZAP 2010-03-17 19:11:32.516233 [CRIT] switch_loadable_module.c:882 Error Loading module /opt/freeswitch/mod/mod_openzap.so **Module load routine returned an error** Is there anything else I need to change when going from zaptel to dahdi? The kernel modules seem to load ok, and /dev/dahdi/* looks good too. -- Brian May <brian(a)microcomaustralia.com.au>

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