April 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Consulting help
by andretodd＠verizon.net 29 Apr '14

29 Apr '14

3 6

hangup with cause:WRONG_CALL_STATE
by rockxsj 29 Apr '14

29 Apr '14

Hi guys! I found that the some of the incoming call will hangup with case: WRONG_CALL_STATE,but else will be ok.After a hour,almost every incoming call will be hangup with the same cause.Then I will restart the freeswitch,it will server like before:some call will be hangup with the cause... 28e10f2e-c851-11e3-a146-85a8fadad1b0 2014-04-20 14:01:15.498169 [WARNING] switch_core_state_machine.c:567 28e10f2e-c851-11e3-a146-85a8fadad1b0 sofia/internal/123456(a)sipsj.236.cn Abandoned 28e10f2e-c851-11e3-a146-85a8fadad1b0 2014-04-20 14:01:15.498169 [NOTICE] switch_core_state_machine.c:570 Hangup sofia/internal/123456(a)sipsj.236.cn [CS_NEW] [WRONG_CALL_STATE] What does Wrong call state mean and what can I do to fix this issue? Everything else is working. I can make outgoing calls and call other extensions. Thanks in advance.

3 7

Yealink T48G and TLS
by Iskren Hadzhinedev 29 Apr '14

29 Apr '14

Hello everyone, Just got a couple of new Yealink T48G phones and I am having a couple of rather weird (at least for me) issues with them. Whenever I enable TLS authentication on the phones, they register with the FreeSWITCH box but there's no media on the outbound calls and I'm unable to get any incoming calls at all. If I switch the authentication protocol to TCP or UDP it's all working good. I tried enabling/disabling SRTP for all 3 protocols and it kept the behaviour consistent. All calls made are local (registered to the same FreeSWITCH box) and only these phones have any issues with calls. I can call someone from the Yealink and then add another person in a 3-way conference. That way the two remote parties can hear eachother, but the Yealink is dead silent. Here's the profile on which all phones are connected: <profile name="local"> <domains> <domain name="all" alias="true" parse="false"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="watchdog-enabled" value="no"/> <param name="watchdog-step-timeout" value="30000"/> <param name="watchdog-event-timeout" value="30000"/> <param name="log-auth-failures" value="true"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="5060"/> <param name="dialplan" value="XML"/> <param name="liberal-dtmf" value="true"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="SILK,OPUS,G722,PCMU,PCMA,GSM"/> <param name="outbound-codec-prefs" value="PCMU,PCMA,GSM"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="local_stream://moh"/> <param name="record-path" value="$${base_dir}/recordings"/> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="true"/> <param name="tls-only" value="false"/> <param name="tls-version" value="tlsv1"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="5061"/> <param name="tls-cert-dir" value="$${base_dir}/conf/ssl"/> <param name="tls-verify-date" value="true"/> <param name="inbound-late-negotiation" value="true"/> <param name="inbound-zrtp-passthru" value="true"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="yes"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="ext-rtp-ip" value="$${local_ip_v4}"/> <param name="ext-sip-ip" value="$${local_ip_v4}"/> <param name="challenge-realm" value="auto_from"/> </settings> </profile> and (due to their sizes) a tport log, a siptrace for an outgoing call from the Yealink and an incoming call (that never rings the phone) with TLS enabled. The whole setup is: FreeSWITCH -- Internet -- NAT Router -- Yealink and Android phone (in different subnets so no direct LAN communication between them) Any thoughts are greatly appreciated. Thanks in advance! Kind regards,-- Iskren Hadzhinedev

2 3

Bypass media mode with "intercept" application
by FSX 28 Apr '14

28 Apr '14

I'm trying to use GV callback feature in FS and having trouble with making it work in bypass media mode. Incoming calls are working well. Call comes to public context, then it's transferred to local domain context. Then it goes via: <action application="set" data="bypass_media=true"/> <action application="bridge" data="..."/> RTP streams directly between remote SIP (caller party) and local SIP client, as it should be by design (calls have better audio quality and don't use recourses of local SIP server). Now, I try to do the same when I use GV callback feature. Local dial plan (as recommended in Google Voice API <https://wiki.freeswitch.org/wiki/Google_Voice_API> ) sets hash with current "uuid" and runs system call to make callback. Then in public context dial plan incoming call is accepted, using "intercept" application with saved "uuid". Exactly as it's recommended. It's working well, but in this case I can't get bypass mode whatever I try to do. * I tried to set "bypass_media=true" before run system call to make callback * I tried to set "bypass_media_after_bridge=true=true" before run system call to make callback * I tried to export those variables from leg A to incoming call (leg B) * I tried to set it before executing "intercept" application on incoming call * I tried to apply global setting to make sure that FS is always using bypass mode ("inbound-proxy-media" set to "true" in "sip_profiles\internal.xml") Nothing works so far... FS always ignores those settings and proxies RTP stream. How to use bypass media mode in this scenario? Thank you in advance -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Bypass-media-mode-with-interc… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

Fwd: internal transfer decreases Max-Forward channel counter
by Rafael Kaihatu 28 Apr '14

28 Apr '14

Hi, issue description: inbound call arrives with Max-Forwards set to 9 (or some low value), after the 9th internal transfer (it executes the transfer command, but the call doesnt leave the freeswitch instance), the call is dropped. I am using a few transfers to "route the call", and then another few to "put the call on hold and play Music on Hold", eventually I need to execute more than 9. should the switch_ivr_session_transfer function (./src/switch_ivr.c:1716 - v1.2) always decrease the SWITCH_MAX_FORWARDS_VARIABLE channel variable ? is there a way to disable this ? Thanks Rafael

2 1

PC Based Caller-id Monitor?
by Sean Devoy 28 Apr '14

28 Apr '14

Hi All, My client has their DID ringing 2 extensions. He says "at night when they are gone I would love to have caller id from their extension (aka did) to decide if I want to pick up the outside call" He does not want his phone to ring from the DID all the time, but wants the caller-id on his phone or PC. I need ideas how to accomplish this. ESL app? The app would not necessarily have to monitor for incoming calls if there is a way to "query" the caller id of an incoming call to a given extension. Thanks, Sean

2 2

Consulting help
by Andre Demattia 28 Apr '14

28 Apr '14

Hi I'm looking for someone to help me with FreeSwitch on Windows using mod managed. I keep getting a exception and I'm pretty sure it's the c code and not the c# code. I see alot of stack hash errors in unknown modules. I've had no luck debugging it. Anyone interested in consulting work to fix this bug? Andre

2 3

Conference verbose-events
by Adam Kelloway 28 Apr '14

28 Apr '14

Hi there, When I set 'verbose-events' in a conference profile, the channel variables only get added to certain conference::maintenance events. In particular, they seem to be absent from at least the events whose Actions are 'conference-create' or 'conference-destroy'. Is this by design? The wiki suggests that this should apply to all conference events. https://wiki.freeswitch.org/wiki/Mod_conference#Conference_Parameters: /verbose-events Maximum verbosity for transcripting. Events related to the conference will send more data. Specifically the events will include all the channel variables on each conference event. / Thanks, Adam

1 0

Check Email remotely.
by Sean Devoy 28 Apr '14

28 Apr '14

Hi All, How does a user check their FS voicemail when out of the office? We have an IVR on incoming calls. Dialing an extension goes to an extension and then its voicemail. We actually set it so that, for example, to go directly to voicemail for ext 20, you can dial 220. But then what? Is there a key to press while leaving voicemail to "auth" and check voicemail? We have pins set. Thanks, Sean

4 4

Polycom NAT issues AGAIN
by Sean Devoy 28 Apr '14

28 Apr '14

Hi, I am back to hate on Polycom phones again. My customer's power was out for 2 days. Now that everything is repaired the Polycom phones are failing NAT traversal AGAIN. The Cisco phones on site are all perfect. The symptoms are repeated "REGISTER" attempts all with CSeq 1. The only issue I see is that in the request the call-id: ends with @<the external ip address>. All the phones that are registering have the local address there. Anyone know how to configure Polycom 335 phones to connect FROM NAT to public addressed FS? My next choice is to go into the router and map specific ports to these 2 crappy phones. If my time was worth minimum wage I could have purchased them 2 new phones of any brand and model at this point. Any CISCO haters want to swap for some VERY LIGHTLY USED (the never frigging work) Polycom 335s? Or other brand phones that actually work through nat? I'll swap in a heartbeat. I kind of like the Polycom's on a local lan without NAT. Thanks, Sean

2 2