- FreeSWITCH-users - FreeSWITCH Mailing Lists

zrtp endpoints have different sas through fs 1.0.4
by Harondel J. Sibble 19 Jan '10

19 Jan '10

I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp capable endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone client, 2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i (symbian s60) and an O2 Xda Flame (windows mobile 5). All 3 endpoints are registered with FS using the default extensions of 1000- 1003 With global_setvar zrtp_secure_media=true the zrtp negotiation between end points happens but the SAS never matches,below is console output for a call between 2 of the endpoints 2009-08-23 14:10:17.643073 [NOTICE] mod_sofia.c:1509 Pre-Answer sofia/internal/1003(a)10.12.13.45! 2009-08-23 14:10:21.257568 [NOTICE] sofia.c:3794 Channel [sofia/internal/sip:1000@10.12.13.166:5062] has been answered 2009-08-23 14:10:21.275521 [NOTICE] switch_ivr_originate.c:2015 Channel [sofia/internal/1003(a)10.12.13.45] has been answered 2009-08-23 14:10:22.232053 [WARNING] mod_sofia.c:810 We were told to use ptime 20 but what they meant to say was 80 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2009-08-23 14:11:34.496118 [NOTICE] sofia.c:322 Hangup sofia/internal/sip:1000@10.12.13.166:5062 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-08-23 14:11:34.512100 [NOTICE] switch_ivr_bridge.c:1016 Hangup sofia/internal/1003(a)10.12.13.45 [CS_EXECUTE] [NORMAL_CLEARING] 2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1086 Session 16 (sofia/internal/sip:1000@10.12.13.166:5062) Ended 2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/sip:1000@10.12.13.166:5062 [CS_DESTROY] 2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1086 Session 15 (sofia/internal/1003(a)10.12.13.45) Ended 2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/1003(a)10.12.13.45 [CS_DESTROY] Of note, with the endpoints registered through the Ekiga sip server, the sas DOES match on both ends. With global_setvar zrtp_secure_media=false, the endpoints can't detect a zrtp peer. Reading the list archives hasn't enlightened me. I see this comment from 2008 http://www.nabble.com/Freeswitch-and-Twinkle-and-ZRTP- td18518140.html#a18518343 On Jul 17, 2008, at 4:23 PM, Michael Jerris wrote: > it should in bypass_media or proxy_media modes. in the other modes we > are in the media path and would not know how to handle the encrypted > packets. > > Mike Is this still relevant? Or is there some other setting not covered here http://wiki.freeswitch.org/wiki/ZRTP to make this work properly? I ask firstly about this in the context of a peer 2 peer zrtp communication between the endpoints, then secondly in the case of FS acting as a trusted middleman as in section 2 here http://www.zfoneproject.com/docs/asterisk/man/html/u_guide.html#passthrough Lastly how does one implement the security enrollment as noted above with FS -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. help(a)pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice)

3 28

Is possible from dialplan to detect if called party is online?
by Fabio Pietrosanti (naif) 19 Jan '10

19 Jan '10

There some condition that let to check if the called party is currently registered and online, before giving the called a return code? We need to manage a condition where the called number is an iPhone and is usually "always offline" and we need to wakeup the voip client with a "push notifcation". But before doing the push notification we should check if the user is already registered/online or not. Don't know which path to follow Fabio

2 1

How do I invite group to join existing call?
by mailinglist 19 Jan '10

19 Jan '10

Hi All I would like to be able to invite a group / global to join an existing call, but how do I accomplish this, can it be done?

3 4

Changes for deb and rpm files.
by Brian West 19 Jan '10

19 Jan '10

Heads up. http://jira.freeswitch.org/browse/FSCONFIG-17 Someone will need to update the configs for the debs and rpm's /b

1 0

Polycom configuration for FreeSWITCH
by Adam Ford 19 Jan '10

19 Jan '10

Does anyone who has successfully implemented Polycom IP301/501s, with FreeSWITCH, have a configuration they can share with me as an example? I have read the Polycom Admin Guide several times and understand what the settings are/do, I am just not sure which FreeSWITCH supports, which it doesn't, and which need special configuration to work with FreeSWITCH. Thanks in advance, -Adam

4 6

Voicemail to email problems
by John 19 Jan '10

19 Jan '10

Voicemail to email gets a broken pipe error. I believe this is happening while writing the file to /tmp? The mail.xxxxxxxxxxxx file is never written. Any ideas? Thanks

1 0

Which port I have to open?
by Michelle Konzack 19 Jan '10

19 Jan '10

Hello *, my FreeSWITCH was in a DMZ but now I have installed it on a dedicated server (TI OMAP3530) and nothing as working... :-/ Please can someone tell me which ports I have to forward from my router fo Freeswitch and reverse? Thanks, Greetings and nice Day/Evening Michelle Konzack Systemadministrator Electronic Engineer Tamay Dogan Network Debian GNU/Linux Consultant -- Linux-User #280138 with the Linux Counter, http://counter.li.org/ ##################### Debian GNU/Linux Consultant ##################### <http://www.tamay-dogan.net/> Michelle Konzack <http://www.can4linux.org/> Apt. 917 <http://www.flexray4linux.org/> 50, rue de Soultz Jabber linux4michelle(a)jabber.ccc.de 67100 Strabourg/France IRC #Debian (irc.icq.com) Tel. DE: +49 177 9351947 ICQ #328449886 Tel. FR: +33 6 61925193

3 4

Error in finding OpenZAP span id
by Magesh R 19 Jan '10

19 Jan '10

Dear All, I have installed Sangoma PRI card and installed wanpipe drivers. The wanrouter process started sucessfully. I had the following configurations, openzap.conf: ========== [span wanpipe PRI_1] name => OpenZAP number => 1 trunk_type => e1 b-channel => 1:1-15 d-channel => 1:16 b-channel => 1:17-31 [span wanpipe PRI_2] name => OpenZAP number => 2 trunk_type => e1 b-channel => 2:1-15 d-channel => 2:16 b-channel => 2:17-31 openzap.conf.xml: =========== <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="1"/> </settings> <libpri_spans> <span name="PRI_1"> <param name="node" value="cpe"/> <param name="switch" value="ni2"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> </libpri_spans> </configuration> When I started the freeswitch I have received the following error, 2010-01-19 12:41:22.693212 [ERR] mod_openzap.c:2039 Error finding OpenZAP span id: name:PRI_1 Any one please tell me a way to solve this problem... Thanks, Mag.

2 1

IVR returning info via event socket
by Earl Pinkerton 19 Jan '10

19 Jan '10

Hi All, I am trying to call an IVR remotely over the event socket port. I am testing using telnet. I call originate (similar to the following): api originate {origination_caller_id_number=8885551111,ignore_early_media=true}sofia/gateway/teliax/18885552222 &ivr(demo_ivr) This works fine (places the call, runs the IVR, accepts DTMF tones to move between menus). My problem is that I would like the remote app to send a PIN as part of the originate call, then have the callie enter the PIN via DTMF and indicate whether there was a match or not back to the calling app (which is a remote web program). We are trying to keep it simple and have the remote app just call the IVR and get the result, without having to control the IVR through asynchronous events. Is there any way for the IVR to check for a match, then somehow send info back over the port to the calling app? By the way, I tried using menu-exec-api inside ivr.conf.xml (to see if I might be able to use the echo api call or something similar), but I got the following error: 2010-01-19 01:22:55.796658 [WARNING] switch_ivr_menu.c:704 Invalid Action [menu-exec-api] Thanks.

1 0

Bria softphone registration issue (NAT/rport issue?)
by Adam Ford 19 Jan '10

19 Jan '10

I don't know if any of you would be able to help with this, but I figured I would ask. After a few hours of inactivity, it seems my Bria softphones lose connection with FreeSWITCH. They are able to call out, but when calling in I just get silence until it goes to voicemail. It sounds like a classic NAT issue, except the NAT is completely handled by a Soincwall that has VoIP features to support SIP transformations (SonicOS Enhanced 5.x). It seems as though after a period of time the port in which the Sonicwall is assigning for the SIP UA and the port that FreeSWITCH has registered get out of sync, despite the Bria re-registering every 5 minutes. Could this be caused by 'Use rport' being enabled by default in the Bria softphone? Setup - FreeSWITCH 1.0.4 Bria 2.5.4 Sonicwall E5500 with SIP transformations and Consistent NAT enabled Thanks in advance for any help, -Adam

3 9