August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

sofia_contact result
by Gerald Weber 12 Aug '12

12 Aug '12

Hi, can i rely on the structure of the result string i get back from e.g.: freeswitch@default> sofia_contact 2000 (or sofia_contact user/2000) sofia/internal/sip:2000@192.168.20.219:5060 thanks gw

2 4

uuid_media off appears to cancel park_after_bridge
by Lynn Nielson 11 Aug '12

11 Aug '12

We are building a dialer interface to our client app that calls our customers and then allows those customers to connect and autodial to their clients. We are using these commands to do this: originate {park_after_bridge=true}sofia/gateway/vitelity-outbound/xxxxxxxxxx &park +OK <uuid_1> originate sofia/gateway/vitelity-outbound/yyyyyyyyyy &park +OK <uuid_2> uuid_bridge <uuid_1> <uuid_2> uuid_media off <uuid_2> If the second call (leg-b) hangs up, then both calls are dropped and the park_after_bridge=true appears to be ignored. If we leave off the last command of uuid_media off <uuid>, then the call behaves as expected on hang-up leaving the first call (leg-a) open. My problem is the we have no audio on the call when we do not add the uuid_media off command. This only happens when bridging two external calls. If one of the legs is an internal extension, then we don't need the uuid_media off and audio works fine. Is there another way of doing this or a way for uuid_media off to honor the park_after_bridge condition? btw: we have also tried this using Flowroute as the gateway as well. Thanks Lynn

1 0

mod_fifo
by Vishal Kakkar 11 Aug '12

11 Aug '12

Hi all, I want to put the caller into fifo queue *without answering the call till agent attend it.* But fifo is answering the call while putting caller in queue. Please help!! Thanks, -Vishal.

5 5

Smack a Hacker context?
by Sean Devoy 11 Aug '12

11 Aug '12

HI Everyone, I didn't bother to obliterate my servers real identity from my last pastebin post, shame on me. Of course within 24 hours some a$$hole is trying the default login ids for Freeswitch. I am not quite a noob here, but this would take me some time to do and I know some of you guys could crank this out in minutes. Of course you all probably have better ideas too. I want to allow the users in the sample Freeswitch config to login - but to special context. In the "special" context, everything they dial plays a recording like "you're an ass hangup and try again" or perhaps and ear piercing LOUD tone. Even better would be to dial them every few minutes with the same recording or sound. Best of all, if we can detect that they are on a CISCO 5xx phone, I have a special config file to send them! Anyone got better ideas? I would love to hear them. I suppose the adult choice would be to gather their external and internal ip addresses and report them to their ISP, but that won't achieve much. Maybe we could build a blacklist and watch for them to connect to the Freeswitch conf call. Then we could all tell them how much we enjoy their efforts. Sean

8 8

high load with ~200 fax calls despite low CPU utilization and low IO wait
by Adam Johnston 11 Aug '12

11 Aug '12

Hi all, My FreeSWITCH setup works as follow: I have a service that launches and monitors faxes (via event sockets) on one of a few FreeSWITCH instances. These FreeSWITCH instances are running on dual-core, 2ghz CentOS 5.8 and CentOS 6.0 VMs. The issue is that once I get to ~200 simultaneous faxes on any one FreeSWITCH VM the load average shoots up (typically ~30, although I've seen it get much higher) and the VM becomes sluggish and occasionally unusable. top and sar output confirms that CPU utilization is pretty low, as are IO wait and RAM usage. I'm using a git head from 2012-02-06, although I see the same problem if update to a more current head. On the most recent head I tried (2012-08-06) the issue is worse, and crashes with a backtrace similar to this Jira, http://jira.freeswitch.org/browse/FS-2893 despite my compiling Lua and including the FreeSWITCH bindings. These issues occur on both the CentOS 5.8 and 6.0 VMs. I'm in the process of loading 5.8 onto a physical machine and testing there. It's also worth nothing that this load issue does not occur when I launch faxes from a script on one of the VMs. Has anyone else seen similar high load issues before? Many thanks, Adam Johnston

7 9

Call Diversion/CF (302 Moved Temporarily) not working
by Johannes Jakob 11 Aug '12

11 Aug '12

Hi, My setup looks like this: User A (1001), User B (1002) and User C (1003) are registered to 13.13.13.66 via OpenSIPs Proxy 222.222.222.222. User B activated unconditional call forwarding on his phone to User C. Now User A (or any external caller) calls User B. Until last week User A would then be able to talk to User C - the expected behaviour. On Saturday, I updated the freeswitch from version 9fe08675a1 to d1c3f910a6 to get and test some of Steve's fax changes. Since this upgrade, Call-Forwarding / Call Diversion is broken on a level, that I don't understand. User A: INVITE 1002 to B's phone => User B: 302 Moved Temporarily (Contact: <sip:1003@sip.isp.net;user=phone>) => FS: ACK => FS: INVITE 1003 to *B*'s phone! => User B: 404 Not Found => FS: ACK => User A gets signalled the 404 Not Found. packet trace below. I'm aware of http://jira.freeswitch.org/browse/FS-724 http://jira.freeswitch.org/browse/FS-821 but those didn't help me solve my issue: internal.xml: <param name="apply-nat-acl" value="nat.auto"/> internal.xml: <param name="aggressive-nat-detection" value="false"/> I added aggressive-nat-detection=false yesterday, before it wasn't in the config at all. I also tried removing the nat.auto acl temporarily, but that didn't help either. Yesterday afternoon I updated to git c3de9637af, did several bootstrappings, cleans and makes... no change. So today I tried to downgrade to 9fe08675a1, the version I had running until Friday and that was working fine... well, it doesn't now. So I can't even _prove_ it was working before :( Can somebody please point me in the right direction? Any hint in the right direction is much appreciated! Thanks! ====================================================================================== SIP Trace (tcpdump): 16:38:58.950321 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472 E....[ .>..;^... .......D.fINVITE sip:1002@137.137.137.245:7390;line=ydwzxzpg SIP/2.0 Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB Max-Forwards: 28 From: "User A" <sip:1001@sip.isp.net>;tag=3r8K7r5Hppe6N To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:mod_sofia@13.13.13.66:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 442 P-Key-Flags: keys="3" X-AUTH-IP: 137.137.137.245 X-FS-Support: update_display,send_info Remote-Party-ID: "User A" <sip:1001@sip.isp.net>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1344322582 1344322583 IN IP4 13.13.13.66 s=FreeSWITCH c=IN IP4 13.13.13.66 t=0 0 m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 13 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 teleph 16:38:58.950328 IP 222.222.222.222 > 172.16.12.210: udp E....[..>.4.^... ..one-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 27936 RTP/AVP 34 102 a=rtpmap:34 H263/90000 a=rtpmap:102 H264/90000 16:38:58.979148 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 623 E.....@.@. . ..^........w8$SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKN30pm01NgUrrB Record-Route: <sip:222.222.222.222;lr;ftag=3r8K7r5Hppe6N;did=d3f.16f40285> From: "User A" <sip:1001@sip.isp.net>;tag=3r8K7r5Hppe6N To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg>;tag=mzey6xfxhz Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:1003@sip.isp.net;user=phone> Diversion: <sip:1002@137.137.137.245:7390;line=ydwzxzpg>;reason="unconditional" Content-Length: 0 16:38:58.979954 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 358 .^....@.>. .......n.fACK sip:1002@137.137.137.245:7390;line=ydwzxzpg SIP/2.0 Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK5cc7.3e8f347.0 From: "User A" <sip:1001@sip.isp.net>;tag=3r8K7r5Hppe6N Call-ID: 76f09f61-5b40-1230-96b6-0016367615cd To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg>;tag=mzey6xfxhz CSeq: 31839105 ACK Max-Forwards: 70 Content-Length: 0 16:38:58.985405 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 1472 E....\ .>..:^... .......ZgIINVITE sip:1003@137.137.137.245:7390;user=phone SIP/2.0 Record-Route: <sip:222.222.222.222;lr;ftag=411c9KpNKZ4rH;did=e81.20343ac4> Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ Max-Forwards: 28 From: "User A" <sip:1001@sip.isp.net>;tag=411c9KpNKZ4rH To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Contact: <sip:mod_sofia@13.13.13.66:5060> User-Agent: FreeSWITCH Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 466 P-Key-Flags: keys="3" X-AUTH-IP: 137.137.137.245 X-FS-Support: update_display,send_info Remote-Party-ID: "User A" <sip:1001@sip.isp.net>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1344322582 1344322584 IN IP4 13.13.13.66 s=FreeSWITCH c=IN IP4 13.13.13.66 t=0 0 m=audio 27756 RTP/AVP 8 0 98 99 9 100 3 101 a=rtpmap:98 G7221/32000 a=fmtp:98 bitrate=48000 a=rtpmap:99 G7221/16000 a=fmtp:99 bitrate=32000 a=rtpmap:100 iLBC/8000 a=fmtp:100 mode=30 a=rtpmap:101 telephone-e 16:38:58.985409 IP 222.222.222.222 > 172.16.12.210: udp E....\..>.4.^... ..vent/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=video 27936 RTP/AVP 34 102 a=rtpmap:34 H263/90000 a=rtpmap:102 H264/90000 16:38:59.031218 IP 172.16.12.210.2054 > 222.222.222.222.5060: SIP, length: 551 E..C..@.@..7 ..^......../..SIP/2.0 404 Not Found Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 Via: SIP/2.0/UDP 13.13.13.66;received=13.13.13.66;rport=5060;branch=z9hG4bKpctFpUjSD4eBQ From: "User A" <sip:1001@sip.isp.net>;tag=411c9KpNKZ4rH To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd CSeq: 31839105 INVITE Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Length: 0 16:38:59.031922 IP 222.222.222.222.5060 > 172.16.12.210.2054: SIP, length: 341 E..q..@.>...^... .......].mACK sip:1003@137.137.137.245:7390;user=phone SIP/2.0 Via: SIP/2.0/UDP 222.222.222.222;branch=z9hG4bK61fb.31c6b792.0 From: "User A" <sip:1001@sip.isp.net>;tag=411c9KpNKZ4rH Call-ID: 76f61af9-5b40-1230-96b6-0016367615cd To: <sip:1002@137.137.137.245:7390;line=ydwzxzpg> CSeq: 31839105 ACK Max-Forwards: 70 Content-Length: 0 ======================================================================================

2 2

Passing DTMF in Conference?
by Magnus 11 Aug '12

11 Aug '12

Hello all, Is it possible to allow dtmf to flow when calls are in conference bridge? For example if one auto connects two users to conference bridge, they are placed into conference automatically, at this point one of the conference users' needs to be able to pass a digit to send as dtmf to the other conference end point device, which they need to detect, alas it seems that the dtmf from user 1 is not bridged to user 2 by default? DTMF used is RF2833 Any tips or thoughts on this welcome. Thanks Magnus

2 1

Error with MAKE CURRENT on SPANDSP
by Guillermo Ruiz Camauer 10 Aug '12

10 Aug '12

On CentOS 6.3, fully updated. Was on FreeSWITCH Version 1.2.0-rc2+git~20120621T145213Z~2091f4f0d3+unclean~20120622T125056Z Started. Just tried to "make current" and it choked when it got to SPANDSP: ... making all mod_spandsp make[5]: Entering directory `/usr/src/freeswitch/src/mod/applications/mod_spandsp' Creating mod_spandsp_la-mod_spandsp.lo quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109: error: expected specifier-qualifier-list before ‘t81_t82_arith_encode_state_t’ /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202: error: expected specifier-qualifier-list before ‘t81_t82_arith_decode_state_t’ In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: error: expected specifier-qualifier-list before ‘lab_params_t’ /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:77: error: expected specifier-qualifier-list before ‘lab_params_t’ In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:112: error: expected specifier-qualifier-list before ‘t42_decode_state_t’ In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:136: error: expected specifier-qualifier-list before ‘t42_encode_state_t’ /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:146: error: expected specifier-qualifier-list before ‘lab_params_t’ In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33: error: expected specifier-qualifier-list before ‘ademco_contactid_report_func_t’ cc1: warnings being treated as errors /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48: error: struct has no members make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 make[5]: Leaving directory `/usr/src/freeswitch/src/mod/applications/mod_spandsp' make[4]: *** [mod_spandsp-all] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 [root@Freeswitch freeswitch]# Any ideas? Anybody else having this problem?

3 4

Send event when FS starting
by Prevail Magid 10 Aug '12

10 Aug '12

Hello, Is it possible to configure event_socket to send event like 'FREESWITCH has been started' on outbound client?

2 1

Re: [Freeswitch-users] DTMF detection giving problems
by Peter Olsson 10 Aug '12

10 Aug '12

Ok. I've never used this function, but it looks like the terminator you have given "#", should be replaced by "0123456789", to make it stop grabbing on first input digit. But as I said - I don't really know :) Mvh Peter Olsson Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För SamyGo Skickat: den 10 augusti 2012 11:17 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] DTMF detection giving problems No, and basically that isn't needed - I only need to capture 1 digit only. and as soon as one digit is captured it can go to next porgram-line. BR Sammy On Fri, Aug 10, 2012 at 1:56 PM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: Did you terminate with the "#" DTMF char? /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För SamyGo Skickat: den 10 augusti 2012 10:38 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] DTMF detection giving problems OK. Fixed that. Now, The thing is when I was testing it I never waited for the menu to end and just pressed buttons normally. And even then I couldn't get any DTMF. On Fri, Aug 10, 2012 at 1:30 PM, SamyGo <govoiper(a)gmail.com<mailto:govoiper@gmail.com>> wrote: is that ms ? God bless the poor callers !! Goodness. wait I'll be back. On Fri, Aug 10, 2012 at 1:23 PM, Peter Olsson <peter.olsson(a)visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: 7 ms seems like a very short timeout to have? /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För SamyGo Skickat: den 10 augusti 2012 10:11 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] DTMF detection giving problems No Ivan , I've no issue at all with DTMFs ever. Like I said there is an Asterisk server which used torun the same IVRs and on same setup and never got such a problem. But not I'm relaying calls from Asterisk to FS and thus I've a SIP peer in between. Calls are coming in OK, I intermittently face this issue where user presses DTMFs and those digits are not recognized by FS-PlayAndGetDigits() I've lines like this in my LUA choice = session:playAndGetDigits(1, 1,1,7, "#",sound_files[1], "", "[0-9]") freeswitch.consoleLog("INFO","User Entered "..choice.." for this Survey\n") Logs for the calls without dtmf here: [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm [INFO] switch_cpp.cpp:1227 User Entered for this Survey [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:1280 --------------------- Another Example--------------------- [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed. [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm [INFO] switch_cpp.cpp:1227 User Entered for this Survey [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 0:960 [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm --------------------- Another Example--------------------- [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm [INFO] switch_cpp.cpp:1227 User Entered for this Survey [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 6:2560 [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm Log for successful DTMF collected is as follows: [DEBUG] switch_rtp.c:3594 Correct ip/port confirmed. [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 1:8800 [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/mymenu.gsm [DEBUG] switch_ivr_play_say.c:2034 Test Regex [1][0-9] [INFO] switch_cpp.cpp:1227 User Entered 1 for this Survey [DEBUG] switch_ivr_play_say.c:1309 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] switch_ivr_play_say.c:1682 done playing file /sounds/thanks.gsm ------- So I can see a pattern here. But I can't figure out this behavior. Is it because I've 1 try only ! and after just once playing of file PAGD() breaks no matter it gets DTMF or not !? Regards, Sammy On Fri, Aug 10, 2012 at 10:09 AM, ivan baydin <runtwistedrat(a)gmail.com<mailto:runtwistedrat@gmail.com>> wrote: Hello Sammy. should check whether the operator is not blocking dtmf I had a situation - in the same statement it worked, on the other - not processed incoming dtmf 2012/8/10 SamyGo <govoiper(a)gmail.com<mailto:govoiper@gmail.com>> Hello All, I've a small lua script which is just an IVR. Calls are landing on Asterisk and then they are relayed over to FS. The problem is that the IVR don't always detects what user pressed. What I've observed is that the DTMF threshold is very high and some people just press slowly/quickly and the tone duration gets low and FS don't detect anything. I've rfc2833 set for the sip trunk between asterisk and freeswitch. I've also liberal-dtmf set in my sofia external.xml profile file which somewhat got me better results but still DTMFs miss sometimes. Please note that the asterisk on which the calls are landing have never faced this issue before. Asterisk has PRI from which it receives the calls and then relay to desired FS server. Regards. Sammy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:5024cf9d32764791410969!

2 1