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Question about the latest FreeSWITCH book
by Jayanth Acharya 30 Aug '13

30 Aug '13

Can anyone having the book let me know if it has material on mod_gsmopen ? It doesn't seem to be a module that many people use, and not sure the book would cover all modules, but it is what I am very interested in. I am looking at the general architecture, design principles, configuration etc. Basically something, that is not there in the Wiki.

9 13

Can't make inbound call to webrtc-registered user (sipml5)
by Alex S 30 Aug '13

30 Aug '13

Hi, i'm trying to make inbound call to webrtc-registered user, which use sipml5 client. But i have an error (USER_NOT_REGISTERED) ======================================= 2013-08-30 17:30:02.016741 [NOTICE] switch_ivr_originate.c:2661 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2013-08-30 17:30:02.016741 [DEBUG] switch_ivr_originate.c:3632 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2013-08-30 17:30:02.016741 [NOTICE] switch_ivr_originate.c:2661 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2013-08-30 17:30:02.016741 [DEBUG] switch_ivr_originate.c:3632 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2013-08-30 17:30:02.016741 [INFO] mod_dptools.c:3194 Originate Failed. Cause: USER_NOT_REGISTERED ======================================= list_users ======================================= fs_user_ext_0YY|ext_incoming|YYY.YYY.YYY.YYY|default|error/user_not_registered||FS User External 0XX|990XX ======================================= I can't see user in users list, BUT i can see him at registrations list sofia status profile INTERFACE reg ======================================= Call-ID: d0xxf526-9zz2-2db0-0yy4-f666abdd4d92 User: 990XX(a)YYY.YYY.YYY.YYY Contact: "FS EXT User 0XX" <sip:990XX@dxxjal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;fs_nat=yes;fs_path=sip%3A990XX%40XXX.XXX.XXX.XXX%3A56251%3Brtcweb-breaker%3Dyes%3Btransport%3Dws> Agent: IM-client/OMA1.0 sipML5-v1.2013.08.10B Status: Registered(WS-NAT)(unknown) EXP(2013-08-30 17:31:02) EXPSECS(246) Host: serverZZZ IP: XXX.XXX.XXX.XXX Port: 56251 Auth-User: 990XX Auth-Realm: YYY.YYY.YYY.YYY MWI-Account: 990XX(a)YYY.YYY.YYY.YYY ======================================= dilaplan ======================================= <context name="ext_group_0XX_outgoing"> <extension name="ext_group_0XX_outgoing" continue="true"> <condition field="destination_number" expression="^(99(0[0-9]{2}))$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="set" data="vm_auto_play=false"/> <action application="set" data="call_timeout=60"/> <action application="set" data="ringback=${ru-ring}"/> <action application="set" data="transfer_ringback=${ru-ring}"/> <action application="set" data="account_id=${user_data(${destination_number}@${domain_name} attr id)}"/> <action application="export" data="sip_invite_to_params=exten=$1"/> <action application="export" data="sip_callee_id_name=$1"/> <action application="export" data="sip_callee_id_number=$1"/> <action application="bridge" data="user/${account_id}@$${location_1}"/> <action application="answer"/> <action application="voicemail" data="default voicemail_1 99$2"/> <action application="hangup"/> </condition> </extension> </context> ======================================= How it possible to make inbound call to this user? With Great Regards -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-inbound-call-to-we… Sent from the freeswitch-users mailing list archive at Nabble.com.

1 0

Random not working?
by David Villasmil 30 Aug '13

30 Aug '13

Hello guys, I've got this in my dialplan: <action application="set" data="rand_val=${expr randomize(&x);ceil(random(0,100,&x))}"/> <action application="log" data="INFO Random dialplan ${expr randomize(&x);ceil(random(0,100,&x))}"/> <action application="log" data="INFO Random value is ${rand_val}"/> and this is what i see in my cli: Dialplan: sofia/external/12345(a)1.2.3.4 Regex (PASS) [respond] destination_number(1234) =~ /^.*/ break=on-false Dialplan: sofia/external/12345(a)1.2.3.4 Action set(rand_val=${expr randomize(&x);ceil(random(0,100,&x))}) Dialplan: sofia/external/12345(a)1.2.3.4 Action log(INFO Random dialplan ${expr randomize(&x);ceil(random(0,100,&x))}) Dialplan: sofia/external/12345(a)1.2.3.4 Action log(INFO Random value is ${rand_val}) 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:167 (sofia/external/12345(a)1.2.3.4) State Change CS_ROUTING -> CS_EXECUTE 2013-08-29 16:40:48.834592 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/12345(a)1.2.3.4 [BREAK] 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:471 (sofia/external/12345(a)1.2.3.4) State ROUTING going to sleep 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:416 (sofia/external/12345(a)1.2.3.4) Running State Change CS_EXECUTE 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:478 (sofia/external/12345(a)1.2.3.4) State EXECUTE 2013-08-29 16:40:48.834592 [DEBUG] mod_sofia.c:230 sofia/external/ 12345(a)1.2.3.4 SOFIA EXECUTE 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:209 sofia/external/12345(a)1.2.3.4 Standard EXECUTE EXECUTE sofia/external/12345(a)1.2.3.4 log(INFO ASR 40 we got ) 2013-08-29 16:40:48.854572 [INFO] mod_dptools.c:1567 ASR 40 we got Why is the "${rand_val}" not being parsed? Thanks, David

4 5

JIRA closes unfixed problems ??
by Jayanth Acharya 30 Aug '13

30 Aug '13

Was browsing through the JIRA database, trying to identify any discussion regarding the audio distortion, and I found several tickets that have been closed (at least, that is how I interpret the closure), because they have been inactive for some time (15 days??). Which effectively means that the bug-closure is not by solving/fixing, but simply timing-out, in the hope that the bug "somehow magically" got fixed in the latest release ?? I hope I am terribly mistaken in my interpretation. J

3 3

Conferencing with SLA
by Bob McCarthy 30 Aug '13

30 Aug '13

I am trying to see if there is a way to keep a conference alive in SLA when the Shared Lines drop out of a call, i.e. call comes in, a shared line answers and then conferences out to a gateway or some other non-shared line using i.e. bgapi expand originate sofia/internal/2001@${somewhere} &sofia_sla(uuid_sharedLine) . (now three channels on the call) Presently when the shared line hangs up the entire call ends. I would like the two or more non-shared entities to remain conferenced. Bob McCarthy

1 0

Get B-Leg Call-ID from Dialplan
by Grant Bagdasarian 30 Aug '13

30 Aug '13

Hello, I'm using the below extension to export the A-Leg Call-ID to the B-Leg. The A-Leg Call-ID is visible in the B-Leg INVITE as the header X-OCallID. <extension name="external_to_internal"> <condition field="destination_number" expression="(\d+)$"> <action application="set" data="hangup_after_bridge=false"/> <action application="set" data="continue_on_fail=true"/> <action application="export" data="sip_h_X-OCallID=${sip_call_id}" /> <action application="bridge" data="sofia/internal/${destination_number}@$${sip_proxy_ip}:$${sip_proxy_port}"/> </condition> </extension> Next to this, I also want to be able to append the Call-ID of the B-Leg to a custom header of the A-Leg. The B-Leg Call-ID would be appended to a response message, like 100, 200 or 4xx, 5xx, 6xx. Is this possible? If so, how? I took a look at using Lua hooks in lua.xml.conf, but I can't get the script to work.

1 1

Roaming Extensions?
by Regan Yelcich 30 Aug '13

30 Aug '13

Can anyone tell me if there's a roaming extension type feature in FreeSwitch? ie. the ability for a user to sit down at any phone, sign in, and that phone becomes theirs. I read about this feature in the FreePBX docs where they describe it as a dynamic user. Thanks.

8 9

Speex codec in low bandwidth mode (narrowband, mode = 8, quality = 1, bit-rate = 3.95 kbps)
by Dmitriy Shumaev 29 Aug '13

29 Aug '13

FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 2013-01-04 15:45:59Z). I'm trying to use speex codec in mode = 8 (quality = 1, bit-rate = 3.95 kbps), but it seems that it only works with 24.6, 42.2, 44.0 kbps (quality = 10). Can FS work with other modes/quality (I'm only interested in mode = 8)? If so, what should I do to configure it to use specific mode / quality (i.e. 8 / 1)? I've modified file <speex.conf.xml> as follows: <configuration name="speex.conf" description="Speex codec configuration"> <settings name="default"> <param name="quality" value="1"/>  <param name="complexity" value="10"/>  <param name="enhancement" value="false"/> <param name="vad" value="false"/> <param name="vbr" value="false"/> <param name="vbr-quality" value="4.0"/> <param name="abr" value="0"/> <param name="dtx" value="false"/> ... </settings> </configuration>

2 1

Detected energy & floor in mod_conference
by Aaron Seelye 29 Aug '13

29 Aug '13

With callers in a conference, if I do a 'conference list' on the fs_cli, I can see the members in a conference and who is talking, but the 'energy detected' field is always at 0, even though the minimum energy is set at 300 (changing the energy threshold doesn't seem to affect detected energy, it's always at 0). Is this something that has gone by the wayside, is slated for implementation later, or am I just reading the wiki wrong? Is it possible to detect the volume at which individual members are talking? Also, what is the significance of 'floor'? All I can find via google is some references to video conferences. Does this have any effect on audio-only conferencing? TIA, -Aaron

1 0

mod_xml_curl load balancing
by Kostenko, Alex 29 Aug '13

29 Aug '13

Hi list, Could somebody tell me mod_xml_curl has ability to make load balancing in last master git or I need to use additional patch for this? I can't find info about this. Thanks.

2 1