April 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

ERROR: switch_rtp.c:389 Failed DTMF payload check.
by Sean Devoy 22 Apr '13

22 Apr '13

HI, I submitted a jira because people get yelled at for not sending a jira on ERRORs. However, I really think this might be a Cisco SPA504G setup issue. I get "[Err] switch_rtp.c:389 Failed DTMF payload check" when I dial OUT and hit an IVR where I have to enter DTMF numbers. My latest was entering CC number, the response I got was "You entered 4", the first of 16 digits I typed! Cisco has some parameters for DTMF and my are set thusly: DTMF Playback Level: 0 DTMF Playback Length: .1 and Inband DTMF Boost: 6dB. Could those values be incorrect or hindering the process? Thanks, Sean

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Can FreeSwitch be used as a Push-to-talk server?
by MrX 22 Apr '13

22 Apr '13

Hello, I have downloaded freeswitch and read through the wiki document, as far as I know FS can behave as a SIP server which enables both call peers speak and listen at the same time. But I wonder can it act as a push-to-talk server that only one peer speak, the others listen, and use a push button to request for speaking? If I can do this, where to start(may be do some configuration or coding)? My best regards,David

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H.264 high profile support
by Jeff Morriss 22 Apr '13

22 Apr '13

Hi all, currently I'm developing a SIP client for Android which is base on the PortSIP VoIP SDK, I want to setup a server in our Lab for test purpose, does freeswtich support H.264 profile? Best regards,

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freeswitch 1.7.0: transport=tcp - doesn't seem to work
by Michel Brabants 22 Apr '13

22 Apr '13

Hello, I'm using freeswitch 1.7.0 and I'm trying to use transport=tcp on one side (big packets) and udp on the other side (extra data is filtered with copy_custom_headers=false). However, when the udp-side (calling side) finished call-setup with sdp, no ack on the invite 200 OK is sent to the tcp-side. Are there problems with tcp or may I be doing something wrong? The tcp-side is using a gateway which is configured with tcp. Thanks, Michel

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Test
by Blake Priddy 22 Apr '13

22 Apr '13

I haven't gotten an email since around 2 am from all of you. -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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show channels question
by Siri MM 22 Apr '13

22 Apr '13

Hello, I have a scenario where I first place an incoming call into a FIFO, and then pop it out to bridge it to another leg. In a different thread in my application, I keep monitoring the channels, to ensure that my own application is in sync with whatever is happening within freeswitch. What I see is, after the call has been removed from the FIFO, and bridged, show channels still reports application as FIFO. This confuses my application, as it thinks there should be a call in the FIFO, but when it calls FIFO count, the count returned is 0. My question is, is there a way for me to differentiate, in the show channels output, if a call is indeed in a FIFO, or if it has already been popped out and bridged? One difference that I see, between the outputs of show channel before and after the bridge, is 'callee_direction' - before the call is bridged, this is generally blank, but once the call has been bridged, it is either SEND or RECV. Can I use this to differentiate between the two? Thanks!

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USER_NOT_REGISTERED
by info＠shishir.com.np 22 Apr '13

22 Apr '13

hi I am using opensips as a registration server and FS as a media proxy. I dont authenticate the users on the FS but opensips does. I am using mod_curl_xml to get the user extensions and group settings. The issue I am having is when I call to the group of the users after getting data back from mod_curl_xml I get USER_NOT_REGISTERED and hangups the call. This is what I receive back from mod_curl_xml <document type="freeswitch/xml"> <section name="directory"> <domain name="c24118.c.XXXXXXX.com"> <groups> <group name="201"> <users> <user id="201"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(users/${dialed_user}@${dialed_domain})}"/> </params> </user> </users> </group> </groups> </domain> </section> </document> This is what I get on the freeswitch 2013-04-21 07:50:37.000717 [DEBUG] switch_event.c:1608 Parsing variable [presence_id]=[201(a)c24118.c.XXXXXXX.com] 2013-04-21 07:50:37.000717 [NOTICE] switch_ivr_originate.c:2640 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2013-04-21 07:50:37.000717 [DEBUG] switch_ivr_originate.c:3606 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2013-04-21 07:50:37.000717 [NOTICE] switch_ivr_originate.c:2640 Cannot create outgoing channel of type [group] cause: [USER_NOT_REGISTERED] 2013-04-21 07:50:37.000717 [DEBUG] switch_ivr_originate.c:3606 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2013-04-21 07:50:37.000717 [INFO] mod_dptools.c:3084 Originate Failed. Cause: USER_NOT_REGISTERED Am I missing anything ? what I am doing wrong here? Thanks in advance

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ftdm console commands not responding as expected
by bedgar＠vseinc.com 22 Apr '13

22 Apr '13

Hi again freetdm Experts! There are many options for ftdm in the FS console: ftdm list ftdm start <span_id|span_name> ftdm stop <span_id|span_name> ftdm reset <span_id|span_name> [<chan_id>] ftdm alarms <span_id> <chan_id> ftdm dump <span_id|span_name> [<chan_id>] ftdm sigstatus get|set <span_id|span_name> [<chan_id>] [<sigstatus>] ftdm trace <path> <span_id|span_name> [<chan_id>] ftdm notrace <span_id|span_name> [<chan_id>] ftdm gains <rxgain> <txgain> <span_id|span_name> [<chan_id>] ftdm dtmf on|off <span_id|span_name> [<chan_id>] ftdm queuesize <rxsize> <txsize> <span_id|span_name> [<chan_id>] ftdm iostats enable|disable|flush|print <span_id|span_name> <chan_id> ftdm ioread <span_id|span_name> <chan_id> [num_times] [interval] ftdm core state [!]<state_name> ftdm core flag [!]<flag-int-value|flag-name> [<span_id|span_name>] [<chan_id>] ftdm core spanflag [!]<flag-int-value|flag-name> [<span_id|span_name>] ftdm core calls If I do a ftdm list I get: freeswitch@internal> ftdm list +OK span: 1 (wp1) type: analog_em physical_status: ok signaling_status: UP chan_count: 24 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options: none I would like to be able to stop this span but get errors trying to do it. freeswitch@internal> ftdm stop wp1 -ERR failure I think I am using the right name because I can restart the span and restart a single channel in the span. freeswitch@internal> ftdm reset wp1 Resetting channel wp1:1 Resetting channel wp1:2 Resetting channel wp1:3 Resetting channel wp1:4 Resetting channel wp1:5 Resetting channel wp1:6 Resetting channel wp1:7 Resetting channel wp1:8 Resetting channel wp1:9 Resetting channel wp1:10 Resetting channel wp1:11 Resetting channel wp1:12 Resetting channel wp1:13 Resetting channel wp1:14 Resetting channel wp1:15 Resetting channel wp1:16 Resetting channel wp1:17 Resetting channel wp1:18 Resetting channel wp1:19 Resetting channel wp1:20 Resetting channel wp1:21 Resetting channel wp1:22 Resetting channel wp1:23 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c1][1:1] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c2][1:2] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c3][1:3] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c4][1:4] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c5][1:5] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c6][1:6] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c7][1:7] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c8][1:8] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c9][1:9] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c10][1:10] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c11][1:11] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c12][1:12] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c13][1:13] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c14][1:14] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c15][1:15] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c16][1:16] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c17][1:17] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c18][1:18] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c19][1:19] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c20][1:20] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c21][1:21] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c22][1:22] VETO state change from DOWN to RESET 2013-04-19 19:32:03.218963 [WARNING] mod_freetdm.c:4663 [s1c23][1:23] VETO state change from DOWN to RESET freeswitch@internal> ftdm reset wp1 1 Resetting channel wp1:1 2013-04-19 19:33:24.838965 [WARNING] mod_freetdm.c:4656 [s1c1][1:1] VETO state change from DOWN to RESET Thanks for looking and helping if you can! Brian C. Edgar, Jr.

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session_locate and deadlocks
by Michel Brabants 21 Apr '13

21 Apr '13

Hello, I'm trying to passthrough some events from one leg of a bridged session to the other leg: notify-talk and hold. talk was an easy one. However, regarding the hold, I'm afraid of deadlocks, because I will be modying the peer-session (putting it on hold), which will also try to modify the originating leg (play hold-music). So, can a session get a readlock on a session that is currenly processing a request? I have the impression that there are higher-level-locks on sip-level. So, my question is: What is the function of the readlock created by "switch_core_session_locate" ? And, can multiple locks be acquired (think so), but what should I watch out for then? Thanks, Michel

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Security considerations for configuration of a live internet system
by Topo Lino 21 Apr '13

21 Apr '13

Alright, I need use external tools as fail2ban and better use TLS, but in configuration... I understand the security configuration is on "sip_profiles". My current configuration is: <param name="auth-calls" value="true"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="log-auth-failures" value="true"/> Is there some other option recommended to use on "sip_profiles"? On the directory. Is it possible use only "user id" for authentication and don't accepts "number-alias" as username? Thanks. -- Localnet <http://www.localnet.org.es/>

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