May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

no ring back mod_managed!
by babak yakhchali 06 May '10

06 May '10

Hi I want to play a file before bridging a call in mod_managed and I'm using execute_on_answer to begin the recording so I can not use it again and I'm using this code: Session.Execute("Playback",m_readyPrompt); Session.SetVariable("ringback", Session.GetVariable("us_ring")); //I've tested this before and after exec of playback but no change Session.Execute("bridge", string.Format("sofia/internal/{0}%{1}", op, Session.GetVariable("domain"))); but no ring back is played Thanks

1 0

outbound sip call via lua: NORMAL_TEMPORARY_FAILURE
by Neil Patel 06 May '10

06 May '10

I was testing a luascript that I am using to initiate calls from the command line using luarun. Here is the relevant snippet of the script: -- make the call session = freeswitch.Session(DIALSTRING_PREFIX .. phone_num) session:setVariable("caller_id_number", phone_num) session:setVariable("playback_terminators", "#"); session:setHangupHook("hangup"); session:setInputCallback("my_cb", "arg"); This was working fine earlier today, but now I can't get the call to work: 2010-05-05 18:05:46.051908 [NOTICE] sofia.c:4789 Hangup sofia/internal/ sip:1001@127.0.0.1:61770 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2010-05-05 18:05:46.053759 [ERR] switch_ivr_originate.c:2430 Cannot create outgoing channel of type [user] cause: [NORMAL_TEMPORARY_FAILURE] 2010-05-05 18:05:46.053759 [ERR] switch_cpp.cpp:604 session is not initalized 2010-05-05 18:05:46.053759 [ERR] switch_cpp.cpp:604 session is not initalized 2010-05-05 18:05:46.053759 [ERR] freeswitch_lua.cpp:194 session is not initalized 2010-05-05 18:05:46.053759 [ERR] freeswitch_lua.cpp:222 session is not initalized I can paste more details, but seeing as this was working a short time ago, I'm wondering whether it is something silly. Thanks in advance. -Neil

2 2

run error after building in vs 2008!
by babak yakhchali 06 May '10

06 May '10

Hi freeswitch is built without any problems but when running it gives the error about "entry point inet_ntop could not be located in dynamic link library WS2_32.dll" I'm using xp sp3. Thanks

9 13

debug DTMF payload
by Madovsky 06 May '10

06 May '10

Hi, is there any settings in FS that can debug DTMF packets ? Or any soft that analyze if it's good or bad DTMF packet. Thanks F

2 2

mod_com_g729 DECODER CREATE FAILED
by Peter P GMX 05 May '10

05 May '10

Hello, I tried mod_com_g720 with 2 licenses and ran into a problem: What DOES work with G729: * Calling Mailbox (Phone is Snom 360) * Calling external Numbers through Patton gatway What does NOT work * Calling another Snom Phone with G.729 enabled (both Phones use TLS/SRTP) Here an exempt from the log calling phone is 200, called phone is Snom 320: 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3632 Audio Codec Compare [G729:18:8000:20]/[G729:18:8000:20] 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2293 Already using G729 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:3565 Set 2833 dtmf send payload to 101 2010-04-28 16:47:57.022827 [DEBUG] sofia.c:4619 Processing updated SDP 2010-04-28 16:47:57.022827 [DEBUG] sofia_glue.c:2580 Audio params are unchanged for sofia/internal/sip:208@192.168.178.126:5060. 2010-04-28 16:47:57.029474 [INFO] mod_com_g729.c:146 DECODER CREATE - 0x9a09510 0x8e774c8 2010-04-28 16:47:57.128943 [ERR] mod_com_g729.c:142 DECODER CREATE FAILED - 0x8e9f9a8 (nil) 2010-04-28 16:47:57.128943 [ERR] switch_core_io.c:327 Codec G.729 decoder error! 2010-04-28 16:47:57.128943 [DEBUG] switch_ivr_bridge.c:478 sofia/internal/200(a)fs00.telefaks.biz ending bridge by request from read function g729_status Permitted G.729AB channels: 2 Encoders in use: 0 Decoders in use: 0 Here is the dialplan <extension name="Local_Extension_no_Voicemail">. <condition field="destination_number" expression="^\S*$" break="never">. <action application="set" data="transfer_ringback=${us-ring}"/>. <action application="set" data="dialed_ext=208"/>. <action application="set" data="effective_caller_id_number=200"/>. <action application="set" data="effective_caller_id_name=Peter "/>. <action application="db" data="insert/call_return/${dialed_ext}/${caller_id_number}"/>. <action application="db" data="insert/last_dial_ext/${dialed_ext}/${uuid}"/>. <action application="set" data="dialed_ext=208"/>. <action application="export" data="dialed_ext=208"/>. <action application="ring_ready" />. <action application="bridge" data="user/208(a)my.domain"/> </condition>. </extension> Just tested it with an Aastra Phone without TLS: Same Behaviour. Anybody has a clue how to solve this? Best regards Peter

3 10

Are Freeswitch epochs strictly unixtime?
by Frank Church 05 May '10

05 May '10

After converting an epoch to a Microsoft date time value I realized that the result was an hour behind. I am in British Summer Time now. I just want to confirm that all epoch values in Freeswitch are independent of time zone. -- Frank Church ======================= http://devblog.brahmancreations.com

2 1

Freeswitch - Polycom IP 300
by C Sarti 05 May '10

05 May '10

Hi I dont know if I should post here for help.. Im new with FS. I have installed FS in win2003 , with softphone so far is working fine, I am trying to use couple of Polycom 300 I had with callcentric (no luck :( ) at the FS console I have anable siptrace and got the following error: SIP/2.0 401 Unauthorized. via: sip/2.0/udp 192.168.0.2;branch=z9hG4bke36814c786B88EF;received= ...... honestly, from here on Im lost :( any idea what to do?thank you in advanced.regardsCarlos _________________________________________________________________ Hotmail is redefining busy with tools for the New Busy. Get more from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:…

2 1

How to make scheduled calls ?
by hung nguyen 05 May '10

05 May '10

Hi list. I want config FS to make automatic calls follow a fixed schedule. Example: In fixed time in the future, FS will call a fixed user and play a fixed music file for this user. I don't want to use crontab to do this because i want to deploy large schedules. Thank to suggestions.

3 2

Sangoma prid and multiple PRIs with FreeSWITCH
by Bryan Vyhmeister 05 May '10

05 May '10

I setup a FreeSWITCH pbx with a Sangoma A102D card, the latest FreeSWITCH as of yesterday, and Wanpipe 3.5.11. I have found that I cannot use both PRI ports with Sangoma's smg_prid. The first one always works great but the second one never works. I get the following error: 2010-05-05 09:42:02.573743 [CRIT] ozmod_sangoma_boost.c:257 SPAN is not online. 2010-05-05 09:42:02.573743 [ERR] switch_ivr_originate.c:2480 Cannot create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION] When I start FreeSWITCH up I get this error on the console right after all the channels are activated. 2010-05-05 07:25:59.414444 [ERR] ozmod_sangoma_boost.c:1534 Error: Opening MCON Socket [-1] Address already in use The configs are below: openzap.conf [span wanpipe 1] name => 1 trunk_type =>t1 b-channel => 1:1-23 [span wanpipe 2] name => 2 trunk_type =>t1 b-channel => 2:1-23 openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="0"/> </settings> <boost_spans> <span id="1"> <param name="node" value="cpe"/> <param name="switch" value="ni2"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> <span id="2"> <param name="node" value="cpe"/> <param name="switch" value="ni2"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </boost_spans> <analog_spans> </analog_spans> </configuration> I'm dialing with this dial string. <action application="bridge" data="{origination_caller_id_number=${outbound_caller_id_number}}openzap/2/a/$1"/> I am certain the rest of this config works fine because the other PRI works great. I have also attempted to use FreeSWITCH without Sangoma's smg_prid by specifying a d-channel in openzap.conf. That seems to work but calls tend to drop frequently. I would rather just use Sangoma's smg_prid. I also attempted a hybrid config where FreeSWITCH worked without smg_prid on the first PRI and the second PRI was handled by smg_prid. That config worked except for the dropped calls that happen with using FreeSWITCH to manage the d-channel. My question is how can I get two PRI's working with Sangoma's smg_prid? Thank you. Bryan

2 2

External SIP profile
by Kurt Ward 05 May '10

05 May '10

Maybe this is a bit of an uneducated noob question (or misunderstanding about SIP) but here goes: If I have an external profile set up and registering, does it essentially behave like a soft phone would behave? What I am running into is incoming calls work fine, but outgoing calls always produce a 404. If I use a soft phone using the same registration I can make calls with no problem (in both scenarios the calls are to/from 4 digit extensions on a Mitel PBX).

2 2