June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Getting MWI from external SIP voicemail server on Cisco 7970G
by Jeremy Moschner 26 Jun '13

26 Jun '13

Hi Everyone I am running an embedded FreeSWITCH installation (on OpenWRT, opkg version 1.0.7-2) and have a single Cisco 7970G that registers fine and can make and recieve calls (most of the time). However, one of my SIP providers (Internode) also offers a voicemail system (useful if I have my phone turned off and means I don't have to run a voicemail module on my installation). What I would like to find out is if it is possible for the 7970G (or indeed any other endpoint) to query/receive notification from Internode's voicemail system and display a MWI (or some sort of notification on the phone) when a message is there. Now, from my reading, I believe that this is accomplished through either SIP NOTIFY or SUBSCRIBE messages, but I am still quite confused about how it all works. I am unsure if FreeSWITCH needs to "relay" this information to the endpoint, or if is configurable via the MWI-Account parameter (ie. the directory user registers to receive these updates directly from the SIP provide). I imagine I'll need to provide a bit more detail, so please let me know where to start.

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Freeswitch Multi-Tenant Dialplan
by Zoltán Szabó 26 Jun '13

26 Jun '13

Hi, I'm trying to set up a multi tenant config with FS. At the moment i configured only one domain, lets say it is mydomain.com This is my mydomain.com.xml in dialplan directory: <?xml version="1.0" encoding="utf-8"?> <include> <context name="mydomain.com"> <extension name="Internal Call"> <condition field="destination_number" expression="^(\d\d\d\d)$"> <action application="bridge" data="sofia/mydomain.com/$1"/> </condition> </extension> </context> </include> Here is my mydomain.com.xml in the directory directory: <include> <domain name="mydomain.com"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> <variable name="default_gateway" value="$${default_provider}"/> <variable name="default_areacode" value="$${default_areacode}"/> <variable name="transfer_fallback_extension" value="operator"/> <variable name="user_context" value="mydomain.com"/> </variables> <groups> <group name="mydomain.com"> <users> <X-PRE-PROCESS cmd="include" data="mydomain.com/*.xml"/> </users> </group> </groups> </domain> </include> The directory/mydomain.com contains two users (i dont configured user_context here because its configured in the previous config already): 6000.xml: <include> <user id="6000"> <params> <param name="password" value="12345"/> <param name="vm-password" value="12345"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="6000"/>  <variable name="effective_caller_id_name" value="Mr. Teszt Elek"/> <variable name="effective_caller_id_number" value="6000"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> 6001.xml: <include> <user id="6001"> <params> <param name="password" value="12345"/> <param name="vm-password" value="12345"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="6001"/>  <variable name="effective_caller_id_name" value="Mekk Elek"/> <variable name="effective_caller_id_number" value="6001"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> I can login with softphones to 6000 and 6001. Now im trying to call from 6000 to 6001 and it seems not working for some reason. First of all every time i call it tries to get dialplan from the public context. I changed the context parameter in the interlan sip profile so it is now looking for the dialplan in my mydomain.com contexts 2013-06-25 01:59:32.470028 [NOTICE] switch_channel.c:968 New Channel sofia/internal/6000@mydomain.com:5060 [ffc81a48-2c77-4797-8f96-fcf291fc6da3] 2013-06-25 01:59:32.650035 [INFO] mod_dialplan_xml.c:557 Processing 6000 <6000>->6001 in context mydomain.com 2013-06-25 01:59:32.650035 [WARNING] mod_sofia.c:4839 Cannot locate registered user 6001(a)mydomain.com 2013-06-25 01:59:32.650035 [NOTICE] mod_sofia.c:5079 Close Channel N/A [CS_NEW] 2013-06-25 01:59:32.650035 [NOTICE] switch_ivr_originate.c:2608 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2013-06-25 01:59:32.650035 [INFO] mod_dptools.c:3055 Originate Failed. Cause: USER_NOT_REGISTERED 2013-06-25 01:59:32.650035 [NOTICE] mod_dptools.c:3175 Hangup sofia/internal/6000@mydomain.com:5060 [CS_EXECUTE] [USER_NOT_REGISTERED] 2013-06-25 01:59:32.650035 [NOTICE] switch_core_session.c:1506 Session 4 (sofia/internal/6000@mydomain.com:5060) Ended 2013-06-25 01:59:32.650035 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/6000@mydomain.com:5060 [CS_DESTROY] By the way my default port is not 5060 as it is written in the log, its a totally different number. I tried with this bridge too: <action application="bridge" data="sofia/internal/$1(a)mydomain.com"/> The error now with this: 2013-06-25 02:04:06.910054 [NOTICE] switch_channel.c:968 New Channel sofia/internal/6000@mydomain.com:5060 [954bab64-1a79-4cd1-bab1-51d0f87410f2] 2013-06-25 02:04:07.090036 [INFO] mod_dialplan_xml.c:557 Processing 6000 <6000>->6001 in context mydomain.com 2013-06-25 02:04:07.090036 [NOTICE] switch_channel.c:968 New Channel sofia/internal/6001(a)mydomain.com [3f59b312-7b62-4757-87c4-ce737ea405b8] 2013-06-25 02:04:07.090036 [NOTICE] sofia.c:6383 Hangup sofia/internal/6001(a)mydomain.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2013-06-25 02:04:07.113421 [INFO] mod_dptools.c:3055 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2013-06-25 02:04:07.113421 [NOTICE] mod_dptools.c:3175 Hangup sofia/internal/6000@mydomain.com:5060 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2013-06-25 02:04:07.160607 [NOTICE] switch_core_session.c:1506 Session 9 (sofia/internal/6001(a)mydomain.com) Ended 2013-06-25 02:04:07.160607 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/6001(a)mydomain.com [CS_DESTROY] 2013-06-25 02:04:07.223801 [NOTICE] switch_core_session.c:1506 Session 8 (sofia/internal/6000@mydomain.com:5060) Ended 2013-06-25 02:04:07.223801 [NOTICE] switch_core_session.c:1510 Close Channel sofia/internal/6000@mydomain.com:5060 [CS_DESTROY] Have two questions. Why is this error happening? From my asterisk experiences maybe i have to create a context where my extensions can receive the calls? Other question is how can i tell the mydomain.com domain to use the mydomain.com.xml dialplan not the public? Thanks, Zoltan

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No sound plays, Firewall, Nat or ...
by Sayyed Mohammad Emami Razavi 26 Jun '13

26 Jun '13

FS all sip signals detected! user registered! he/she can call 5000 (demo-ivr) and ... but i have no sound from place #1(even-though call starts and demo-ivr starts) and have sound from another place #2! #1 NAT Scenario (My Company Network) FS <> Internet <> NAT <> NAT <> More NAT or Firewall is probable <> SIP Phone #2 NAT Scenario (Wimax wireless internet solutions) FS <> Internet <> Wimax Router NAT <> SIP Phone FS logs when being passed dial plans and starts to play IVR for the scenario #1 with no sound at client: http://pastebin.freeswitch.org/21109 My OS(Fedora) tcpdump logs: http://pastebin.freeswitch.org/21110 any suggestion?!

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forward call External Profile to Internal Profile
by Abdullah 26 Jun '13

26 Jun '13

hi , Hope you are fine , Please help me i want to handle all external request to Internal Profile . mean that if Any Request is coming form DID 234XXX it match with Internal Context Expression. any idea ? THANKS -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/forward-call-External-Profile… Sent from the freeswitch-users mailing list archive at Nabble.com.

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"Stable" branch
by Lesley Pervis 26 Jun '13

26 Jun '13

In fisheye, I compare master to v1.2.stable, and it seems like any changes to master also make it to stable, plus a few more. Also, version on v1.2.stable is 1.2.10, last announced was 1.2.9. Just wondering what's the theory.

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Popular carriers
by Richard Mace 25 Jun '13

25 Jun '13

Hi All, I am putting together some thoughts for a GUI app to configure FreeSWITCH and am wondering what the most popular carriers are to include them into my app. Any thoughts? Richard

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DTMF tones not completely sent
by Oleg Stolyar 25 Jun '13

25 Jun '13

Hi guys, I have a strange problem. I am trying to use the uuid_send_dtmf command to send DTMF tones to a call leg initiated by Level3 (my PSTN connection provider). The tones are *8954. This is a signal to the provider to transfer this call to another PSTN destination. It seems that the provider receives *8 but nothing after that. I tried putting delays between the digits with a "w" and it did not help. After *8 nothing is sent. When the call is connected to another call and the other caller manually presses the same digits, everything works fine. But for some reason it seems that FreeSWITCH command stops sending after the first 2 digits. In the sip profile dtmf type is set to rfc2833 <param name="dtmf-type" value="rfc2833"/> Any ideas? Thank you *Oleg*

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What module do I use to send text message to cellphones ( United States ) ?
by Mahendra Bhegade 25 Jun '13

25 Jun '13

Hi, What module do I use to send text message to cellphone ? Is it GSM_Open or mod_sms. I plan to use lua script to send these text. Any guidance appreciated. Mahendra Bhegade

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Inbound call via SIP URI -- Caller ID
by Tom Carlson 25 Jun '13

25 Jun '13

TLDR; How to you change the domain that is displayed in inbound sip uri caller-id? When somebody calls me at sip:tom@mydomain.com:5080, the call is transferred to my extension, where it rings through on my Nexus 4, which is registered to that extension. If somebody at 1004(a)remote.org calls me, my caller id is showing 1004(a)mydomain.com, which is incorrect. They are not registered on my system, so I would prefer to see 1004(a)remote.org on my caller id. What variable would I use to set the domain portion of the caller id? Modifying name or number, as shown below, simply results in test_number(a)mydomain.com How do I get it to quit displaying mydomain.com and start displaying the domain of the incoming caller? <extension name="toms public sip uri"> <condition field="destination_number" expression="^tom$"> <action application="set" data="effective_caller_id_name=test_name"/> <action application="set" data="effective_caller_id_number=test_number"/> <action application="transfer" data="1000 XML default"/> </condition> </extension>

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Modem Code - How to write a RTP Data handler
by lists＠collisters.com 25 Jun '13

25 Jun '13

I didn't get any response to my query about implementing a modem on freeswitch so I thought I would try to write one. Excuse my ignorance/naivety but can anyone point me in the right direction for writing a module that can receive the RTP frames and decode them using my tone detection algorithm. and for taking my data and sending it back out via RTP frames/packets. Could I take an existing module, and use that as a template, I'm a good C coder but might struggle to start from scratch with FS in my timescales. Any thoughts on where best to start this? All the best, Paul Collister paul(a)collisters.com

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