March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Increase volume of "say" phrases to match play-file?
by Robert Hadley 19 Mar '10

19 Mar '10

Hello, Is there a way to increase the volume of the "say" phrases to match the "play-file" phrases? I am using mod_directory, the sound files from http://svn.freeswitch.org/svn/sounds/trunk/en/ca/june/48000/directory/, and the built-in phrases from /usr/local/freeswitch/conf/lang/en/dir/sounds.xml. Thanks, Robert

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dial by name directory // mod_directory?
by William Traenkle 19 Mar '10

19 Mar '10

Has freeswitch implemented the "Dial by Name" feature where a caller can enter the first x number of characters of an extensions first or last name and have the operator either connect the user to the result and/or state the result's extension number? I heard this was a feature that was implemented or going to be implemented in freeswitch. If it has been implemented, would you send me info on how to configure it? If it has not been implemented, would you state when it is planned to be implemented? I believe the module name is mod_directory, is that correct? Thanks, -Will

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registering with auth id
by srinivasula reddy 19 Mar '10

19 Mar '10

HI, i have a user 1002 in freeswitch, i am trying to register 2001 with help of 1002, mean with auth id is 1002 for me, is it possible to register in this way -- Srinivasula Reddy K

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mod_voicemail modifications
by Tim St. Pierre 19 Mar '10

19 Mar '10

Anybody out there know the inner workings of the voice mail module? It looks like I may have to modify the source code to make it do what I want. It's so close, but I can't quite get it to work. I need to change it so that the voicemail_greeting_number channel variable is used to set the greeting path, but is NOT stored back in the sql database. I would also like to change it so that choosing the option to e-mail the message does not mark it as deleted. I imagine I will get many angry phone calls if I put it into production this way. The prompt doesn't tell you that the message will be deleted if you e-mail it, and not all of my users have e-mail set up correctly. I would prefer messages not to be deleted unless delete is explicitly pressed. I'm pretty new to programming, but I would like to give it a try (unless anyone else out there knows a better way). Can someone point me in the right direction? I mostly want to know where the sql gets updated when the variable is set. That wasn't obvious when I looked through the code. In terms of the e-mail causing the deletion, it looks like the e-mail routine is run in-line with the delete routine, so I'm not sure if they can be separated easily. Thanks for any help anyone can provide. -Tim

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Originate timeout with multiple destinations
by Nicolas Brenner 19 Mar '10

19 Mar '10

Hi, I need to do an originate to multiple destinations, but I need that the first call has a timeout of 17 seconds, and if it's not answered, then call the other destinations. I was thinking something like: originate {ignore_early_media=true}[timeout=17]sofia/gateway/mygateway/xxxxxxx, sofia/gateway/mygateway/yyyyyyyy, sofia/gateway/mygateway/zzzzzz I tried using the call_timeout, timeout and originate_timeout variables in the square brackets for the first dialstring, but it didn't work. Is there a way of doing this with 1 originate command? Thanks!

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Conference with FSComm
by Suneel Papineni 19 Mar '10

19 Mar '10

Hi, I am trying to achieve 3-way (or 3-party) conference scenario using FSComm. Scenario is as follows: An incoming call comes to FSComm from an extension 'A' and is answered. Now FSComm makes a call to another extension 'B' and is answered. Then I want to bridge all three calls and be in conference. So all three parties A, B and FSComm will be in conference. Could someone please give me an idea how can I achieve this scenario using Event Sockets. (want to test with fs_cli using commands). Thanks & Regards Suneel ************************************************************************* Please consider the environment before printing this e-mail ************************************************************************* This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please notify the system manager. http://www.mettoni.com Mettoni Ltd Registered in England and Wales: 4485956 9400 Garsington Road, Oxford Business Park, Oxford, OX4 2HN *************************************************************************

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