August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

enterprise deployment not working
by vip killa 20 Aug '11

20 Aug '11

I'm following tutorial at http://wiki.freeswitch.org/wiki/Enterprise_deployment_<http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> OpenSIPS <http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS> and opensips is not forwarding the registration requests... Running opensips-1.6.4-2-tls, Here is what the debugging looks like when i try to register with the opensip box 10.20.30.17 is the opensips box. 10.20.30.18 is the FreeSWITCH box. Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_msg: SIP Request: Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_msg: method: <REGISTER> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_msg: uri: <sip:10.20.30.17> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_msg: version: <SIP/2.0> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_headers: flags=2 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:get_hdr_field: cseq <CSeq>: <43> <REGISTER> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bKda029da0-e604-1910-80bc-002622a67db9>; state=6 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_via_param: found param type 235, <rport> = <n/a>; state=17 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_via: end of header reached, state=5 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_headers: via found, flags=2 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_headers: this is the first via Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:receive_msg: After parse_msg... Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:receive_msg: preparing to run routing scripts... Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_headers: flags=100 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to: end of header reached, state=10 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to: display={}, ruri={sip:1000@10.20.30.17} Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:get_hdr_field: <To> [24]; uri=[sip:1000@10.20.30.17] ] 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:get_hdr_field: to body [<sip:1000@10.20.30.17> Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:get_hdr_field: content_length=0 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:maxfwd:is_maxfwd_present: value = 70 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:uri:has_totag: no totag Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to_param: tag=70ea9ca0-e604-1910-80bc-002622a67db9 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to: end of header reached, state=29 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_to: display={}, ruri={sip:1000@10.20.30.17} Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:dispatcher:ds_select_dst: set [1] Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:dispatcher:ds_select_dst: alg hash [1], id [1] Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:parse_headers: flags=ffffffffffffffff Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:get_hdr_field: found end of header Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:check_ip_address: params 72.237.213.162, 72.237.213.162, 0 Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:destroy_avp_list: destroying list (nil) Aug 16 13:40:14 voipdev1 /usr/local/sbin/opensips[2580]: DBG:core:receive_msg: cleaning up

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mod_dingaling delay buildup
by Jon 20 Aug '11

20 Aug '11

I am facing delay problems with mod_dingaling. The voice delay increases each minute that I am on an outgoing call. It starts fine and then will get up to 5-10 seconds within 3 minutes. The delay only occurs on the voice of the called party. I have googled and read countless threads and entries. I am running freeswitch on a VM. I have run my setup on CentOS 5.6 i386, CentOS 5.6 x86_64, CentOS 6 x86_64. I have tried using a stun server as well as port forwarding from my router. I have tried using the rtp-autoflush=true and rtp-timer-name=none settings. All of my changes have given me the same results. I am using the current GIT version. I have seen a number of people have a flawless experience with this, so what am I doing differently? Any help? Here are my current jingle configurations: <extension name="gvoice_out"> <condition field="destination_number" expression="^1(\d{10})$"> <action application="set" data="rtp_autoflush=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="ring_ready"/> <action application="bridge" data="dingaling/gtalk/+1$1(a)voice.google.com"/> </condition> </extension> <profile type="client"> <param name="name" value="gtalk"/> <param name="login" value="<email>@gmail.com/talk"/> <param name="password" value="***"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="message" value="FreeSwitch for everybody"/> <param name="rtp-ip" value="auto"/> <param name="auto-login" value="true"/> <param name="sasl" value="plain"/> <param name="server" value="talk.google.com"/> <param name="tls" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="1000"/> <param name="vad" value="both"/>  <param name="candidate-acl" value="wan.auto"/> <param name="local-network-acl" value="localnet.auto"/> <param name="rtp-timer-name value="none"/> <param name="rtp-autoflush" value="true"/> </profile>

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Returned mail: see transcript for details
by johnd＠defyne.org 20 Aug '11

20 Aug '11

The message was not delivered due to the following reason: Your message could not be delivered because the destination server was not reachable within the allowed queue period. The amount of time a message is queued before it is returned depends on local configura- tion parameters. Most likely there is a network problem that prevented delivery, but it is also possible that the computer is turned off, or does not have a mail system running right now. Your message was not delivered within 6 days: Host 106.218.121.36 is not responding. The following recipients could not receive this message: <freeswitch-users(a)lists.freeswitch.org> Please reply to postmaster(a)lists.freeswitch.org if you feel this message to be in error.

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ClueCon 2011 - Assistance w/ posting videos
by Michael Collins 19 Aug '11

19 Aug '11

Hello all! I am looking for a volunteer to assist me with some Drupal/Web work on cluecon.com. I would like to be ready to post the videos as they become available and to do that I need to do some layout work on the cluecon site. Anyone with any kind of CMS/Drupal/Web dev skills would be a perfect candidate to help. If you're in a position to assist please contact me off list. Thanks! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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FreeSWITCH Janitorial Items
by Michael Collins 19 Aug '11

19 Aug '11

Hello all! We have need of your skills with a few sub-projects. Sorry, nothing glamorous, just some behind-the-scenes work. First off, I need to know if anyone out there has experience with UberCart for Drupal. If so, please contact me off list so we can discuss the items I need help with. Secondly, I have a housekeeping task. We have some new "standard vocabulary" files for Callie US English. We need to add them to phrase_en.xml. However, there are quite a few of them and some of them are the same recording with a different inflection. For example, the inflection on the words "one" and "message" are different in these phrases: "To leave a message, press 1." "Press 1 to leave a message." I have a list of all the file names and what's in them but there's a bit of work that needs to be done to document the inflection for each phrase. We also need to settle on a naming convention. If you are in a position to assist with this please contact me off list and I will give you more information. Thanks! -Michael

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Fetching user-variables
by Christian Benke 19 Aug '11

19 Aug '11

Hi! I'm working on a redirect script and ran into a problem with toll_allow: A tries to reach B but B has redirected to C B is allowed to make external calls(via the user-specific-variable "toll_allow"), but A is not(Could be a external caller or a directory-user with restrictions). B's redirection will not work as she intended, as the dialplan is restricted through toll_allow. B is responsible for the external call of A to C, therefore her accountcode and her toll_allow should apply - unfortunately the dialplan doesn't know that the call has been redirected by B. Right now i don't know how to solve this with FreeSWITCH. Is there a way to fetch user-specific-variables of B, like "toll_allow", in a call that was not directly initiated by B? Best regards Christian

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Newbie - external originating leg has no sound when calling
by Charles Bujold 19 Aug '11

19 Aug '11

Weird situation, we have an external location which if they originate the call we cannot hear them, but if we originate the call to them everything works fine. The configuration we have is Sense as firewall that sends and has all stipulated Freeswitch ports open and forwards incoming ports to the Freeswitch server (Ubuntu/Freeswitch/fusionpbx) Internal to internal call works Internal call to the external location works Both parties can talk without issues. External location to Internal location: we do not hear them, but they can hear us. External location call to external location : the originating caller cannot be heard, the second leg works perfectly. (This suggest that it is not a firewall issue but a configuration issue.) External location calls to the voicemail and can record without problems a recording. The problem seems to be limited to the first leg of the call, in that if it originates from an external location the first leg cannot be heard. Any suggestions on how to fix would be appreciated. Thanks! Cjb

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Freetdm bridge to SIP DTMF problem
by Jakub Ouhrabka 19 Aug '11

19 Aug '11

Hi, we've changed OpenZAP to FreeTDM sometime ago and from that time we're experiencing problems with DTMF. We're bridging incoming ISDN30 calls from Sangoma A108DE card + libri + FreeTDM to SIP outgoing calls. Freeswitch is recognizing DTMF correctly but DTMF is not send to outgoing SIP call - it's not present in RTP stream trace (we've setup RFC2833 DTMF signalling). When bridging the call back to ISDN30 DTMF works correctly. We're using latest versions of all software as of 1 month ago. Below are attached two snippets of logs: first is bridging back to ISDN30 where DTMF is sent to called party. Second is bridging to SIP where DTMF is not sent to the called party. We've tried it with both <action application="start_dtmf" /> on and off with no success. Any pointers how to investigate the issue? Thanks, Jakub Log snippets: ISDN30 to ISDN30 bridge - DTMF ok 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1545 [s1c11][1:11] read wanpipe event 3 2011-08-18 12:31:45.950158 [DEBUG] ftmod_wanpipe.c:1415 [s1c11][1:11] Queuing wanpipe DTMF: 5 2011-08-18 12:31:45.950158 [DEBUG] ftdm_io.c:3504 [s1c11][1:11] Queuing DTMF 5 (debug = 0) 2011-08-18 12:31:45.950158 [DEBUG] mod_freetdm.c:733 Queuing DTMF [5] in channel FreeTDM/1:11/XXXXXXXXX 2011-08-18 12:31:45.970623 [DEBUG] switch_ivr_bridge.c:391 Send signal FreeTDM/1:5/XXXXXXXXX [BREAK] 2011-08-18 12:31:45.991084 [DEBUG] ftdm_io.c:3694 [s1c5][1:5] Generating DTMF [5] ISDN30 to SIP bridge - DTMF not forwarded 2011-08-18 12:34:05.287480 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29] read wanpipe event 3 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1545 [s1c29][1:29] read wanpipe event 3 2011-08-18 12:34:05.607708 [DEBUG] ftmod_wanpipe.c:1415 [s1c29][1:29] Queuing wanpipe DTMF: 6 2011-08-18 12:34:05.607708 [DEBUG] ftdm_io.c:3504 [s1c29][1:29] Queuing DTMF 6 (debug = 0) 2011-08-18 12:34:05.607708 [DEBUG] ftmod_libpri.c:1586 -- Caught Event span 1 18 (KEYPAD_DIGIT) 2011-08-18 12:34:05.628171 [DEBUG] mod_freetdm.c:733 Queuing DTMF [6] in channel FreeTDM/1:29/XXXXXXXXXXXXXXX 2011-08-18 12:34:05.628171 [DEBUG] switch_ivr_bridge.c:391 Send signal sofia/external/XXXXXXX@XXXXXXX:5060 [BREAK]

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INFO: Wiki update
by Michael Collins 19 Aug '11

19 Aug '11

Hello all, We have been experimenting with some access controls on the wiki. If you get a dialog box asking for a password then use the same credentials as you would for pastebin.freeswitch.org. You should only see the dialog box when logging in or editing pages. Just searching the wiki does not cause the dialog to show up. We are trying to cut down on the spammers. Let me know if you run into any weirdness. Thanks, MC

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Cannot transfer incoming call
by Tomasz Hyziak 19 Aug '11

19 Aug '11

Hi Transfers from local to local user work without problem, but when a person call in via sip provider (selects call type via ivr and he is put into fifo) and somebody answers this call, he is unable to transfer this call to another person (via *1/*3/*4). When he presses *, he is disconnected. Any ideas why ? 2011-08-19 14:55:17.475054 [DEBUG] switch_rtp.c:3303 RTP RECV DTMF *:800 2011-08-19 14:55:17.475054 [DEBUG] switch_channel.c:2775 (sofia/external/693xxxxxx(a)trunk.dialog.pl) Callstate Change ACTIVE -> HANGUP -- pozdrawiam - Tomasz Hyziak

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