August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to configure zaptel
by Евгений Золотов 29 Aug '08

29 Aug '08

We reinstall FreeSWITCH version to be in the mainstream of project. The last release successfully working for us was 9235 (11.08) we have installed 9377 (28.08) , and have some questions. 1. As we before installation of new release need to have successfully working it would be desirable to have simultaneously installed 2 releases, we do it so: - we rename old (9235) installation to /usr/local/fs.9235; - install new (9377) and rename it also to /usr/local/fs.9377; - for working with necessary version we set link fron directory /usr/local : $ ln -s fs.9377 freeswitch - Seems it's ОК. Is it correct??? 2. We experiment with E1 PRI: > version FreeSwitch Version 1.0.trunk (9377) > load mod_openzap ... API CALL [load(mod_openzap)] output: +OK > oz list API CALL [oz(list)] output: +OK span: 1 type: isdn chan_count: 31 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options none +OK span: 2 type: isdn chan_count: 31 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options none - E1 PRI works correctly (make calls) with same FreeSWITCH (that's 9235 version), we had no time to check up 9377 version yet. Our configuration files below - here all is correct: - /etc/zaptel.conf loadzone = us defaultzone=us span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,1,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 - /etc/openzap/openzap.conf [span zt] name => OpenZAP number => 1 trunk_type => E1 b-channel => 1-15 d-channel => 16 b-channel => 17-31 [span zt] name => OpenZAP number => 2 trunk_type => E1 b-channel => 32-46 d-channel => 47 b-channel => 48-62 - other files /etc/openzap/ didn't change; - we didn't find "trunk_type => E1" in examples of FS, but we find it in examples at www.sangoma.com ; if to change it for "trunk_type => t1" that the result does not change in any way . - /usr/local/fs.9235/conf/autoload_configs/openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="1"/> </settings> <pri_spans> <span id="1">  <param name="mode" value="user" /> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> <span id="2">  <param name="mode" value="user" /> <param name="dialplan" value="XML"/> <param name="context" value="public"/> </span> </pri_spans> </configuration> - which another values (except "user") can be for <param name="mode" value="user" /> ???

2 1

Playback loop and Playback on Parked Call
by Doug Blacksone 29 Aug '08

29 Aug '08

Hi, There are two issues that I am trying to resolve. Is there anyway to continuously loop an audio file? With the playback app, the call gets hung up after the audio file finishes playing, but I need it to replay from the beginning until the caller hangs up. The other issue has to do with the Park app. Is there anyway to playback an audio file ( on a continuous basis) for a parked call? I checked the wiki but can't find any info on that. Any ideas on how to resolve those problems? Thanks, Doug

4 7

CDR accounting question
by Woody Dickson 29 Aug '08

29 Aug '08

Hi, I have a qestion about how to extract the time duration for each individual outgoing call that Freeswitch makes for an incoming call. Here is the scenario : User 10000 --> Freeswitch -------> sofia/voip-gw/160454534333 | -----> sofia/voip-gw2/17864333431 >From Freeswitch's CDR XML output, I can extract the entire duration of the call from the perspective of User 10000 calling Freeswitch. Is there anyway I can tell the duration for each of the two outgoing calls that Freeswitch makes on behalf of User 10000. The reason is because the cost for each one of the outgoing number can be different and I need to correctly account for that. Is it possible for Freeswich to give me that kind of information? Thanks, Woody

2 1

bind meta session suddenly VERY slow response
by Lee JJ 29 Aug '08

29 Aug '08

Hello : I have 2 tested bind_meta_app , after some testing , I found the app either no response or very very slow. Got the [WARNING] already broadcasting 2008-08-29 16:52:22 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1529->1526@inter2 2008-08-29 16:52:22 [INFO] switch_ivr_async.c:1474 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 3 execute_extension::att_xfer XML inter2 2008-08-29 16:52:22 [INFO] switch_ivr_async.c:1474 switch_ivr_bind_dtmf_meta_session() Bound A-Leg: 4 javascript::calltest.js 2008-08-29 16:52:29 [WARNING] switch_ivr_async.c:1983 switch_ivr_broadcast() Channel [sofia/inter2/1529@210.243.126.72][execute_extension::att_xfer XML inter2] already broadcasting...broadcast aborted 2008-08-29 16:52:36 [WARNING] switch_ivr_async.c:1983 switch_ivr_broadcast() Channel [sofia/inter2/1529@210.243.126.72][javascript::calltest.js] already broadcasting...broadcast aborted At first suspecting the DTMF problem , change the rfc2833 and sip info back and forth ... but in vain . freeswitch@localhost> sofia status profile internal CODECS G722,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 check the sip profiles setting <param name="rfc2833-pt" value="101"/>  BTW, where is the in-band DTMF setting ? thanks !

2 1

bind_meta_app for PARK
by Barray McKee 29 Aug '08

29 Aug '08

Hi, Is there anyway that I can set bind_meta_app for PARKed call? When a call is in PARK mode, I want to be able to intercept the DTMP sent so I can do things like: Press 1# to play weather report Press 2# to play music I want to let user to be able to do "things" while the call is on-hold waiting for another party to bridge to it. Can this be done? Regards, Barray

1 0

Unable to execute Playback with event
by Doug Blacksone 29 Aug '08

29 Aug '08

Hi, I am trying to use event to trigger playback on a parked call and I am having difficulty getting it to work: I am not sure what is wrong with this one. Freeswitch responses with ERR: *SendMsg ae67060e-75e5-11dd-9228-efa2040a932a call-command: execute execute-app-name: playback execute-app-arg: /usr/local/freeswitch/sounds/music/8000/suite-espanola-op-47-leyenda.wav lead-frames: 5 loops: 1* Content-Type: command/reply Reply-Text: -ERR invalid session id [ae67060e-75e5-11dd-9228-efa2040a932a ] If I just simply send in execute hangup, it is fine: *SendMsg ae67060e-75e5-11dd-9228-efa2040a932a call-command: execute execute-app-name: hangup *I am sure my UUID is right as I already checked with the following:* freeswitch@ser> show channels API CALL [show(channels)] output: uuid,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate ae67060e-75e5-11dd-9228-efa2040a932a,2008-08-30 00:15:22,1220026522,sofia/internal/1005@192.168.1.104,CS_EXECUTE,1005,1005, 192.168.1.104,1,park,,XML,public,PCMU,8000,PCMU,8000 Content-Type: command/reply Reply-Text: +OK* Any idea what is wrong? Thanks, Doug

1 0

request uri
by jocke eriksson 29 Aug '08

29 Aug '08

Hello! Freeswitch users i hava a simple question, how can i obtain a request parameter in my application, i use mod_java. Like this sip:name@address ;param=value Regards Jocke

2 2

Newbie question
by Sunil Singh 29 Aug '08

29 Aug '08

Hi, I am a newbie in this list and I am finding openser a lot along with freeswitch and asterisk. First what is the use of openser with free switch/asterisk, is it required with free switch if yes in what kind of environment. It helps in scaling the solution or if natting than its not clear. The purpose is not clear. Thanks & Regards, Sunil

2 2

Do we have talk_timeout?
by Adeel Ansari 29 Aug '08

29 Aug '08

Hi all, My concern is to terminate a call using a specific timeout value. I have referred to the docs, what I found is, *originate_timeout*, * progress_timeout*, and *call_timeout*. Do we have something like * talk_timeout*? or anything for similar purpose? Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

4 5

Default voicemail file path
by Juan Backson 29 Aug '08

29 Aug '08

Hi, Can the default voicemail file path be configured to another location? Any hint on where to look would be appreciated. Thanks, JB

2 1