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Cause IE in Progress Message (Sangoma, LibPRI)
by sdemers＠q4innovation.com 26 Aug '10

26 Aug '10

Hi everyone, I'm trying to catch the CAUSE IE into the ISDN NI2 Progress Message with Sangoma A108 card using LibPRI. Based on the Log bellow, the OZMOD_libpri catch the CAUSE IE, but at the event_socket, there is no element with that specification. This is very important because the most carrier doesn't send (90% of the time) the SIT tone UNALLOCATED, and also return NORMAL_CLEARING cause in the disconnect. They send a progress message with an Unallocated (unassigned) Cause IE and play a message on the voice path for a end user. When you have a machine on the other side it's not easy to catch. I've take a look into the ozmod_libpri.c and I didn't found any code related to LPWRAP_PRI_EVENT_PROGRESS. The print-out come probably from LPWRAP_PRI_EVENT_ANY (on_anything) function. Is there a way to push it to the event_socket remote application? Thanks for your answer. ---------------------------------------------------------------------- 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 Handling message for SAPI/TEI=0/0 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- ACKing all packets from 4 to (but not including) 5 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Since there was nothing left, stopping T200 counter 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Stopping T203 counter since we got an ACK 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Nothing left, starting T203 counter 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Protocol Discriminator: Q.931 (8) len=13 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Call Ref: len= 2 (reference 3/0x3) (Terminator) 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Message type: PROGRESS (3) 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < [08 02 80 81] *2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0)* *2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]* 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < [1e 02 80 88] 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 < Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] *2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Processing IE 8 (cs0, Cause)* 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Processing IE 30 (cs0, Progress Indicator) 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 Sending Receiver Ready (6) 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 > [ 02 01 01 0c ] 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 > Supervisory frame: 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 > SAPI: 00 C/R: 1 EA: 0 > TEI: 000 EA: 1 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 > Zero: 0 S: 0 01: 1 [ RR (receive ready) ] > N(R): 006 P/F: 0 > 0 bytes of data 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:106 -- Restarting T203 timer 2010-08-26 11:01:08.849421 [DEBUG] ozmod_libpri.c:979 Caught Event span 1 17 (PROGRESS)

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Easy question, I hope just need to proxy port 5060 to 5080
by Michael Scheidell 26 Aug '10

26 Aug '10

I am in need of putting a sbc or sip proxy in front of our existing sip solution to help solve a very specific problem. The SIP Trunk provider won't send to anything but port 5060, and the existing sip solution listens to port 5080 ONLY. (background: sipxecs. yes, they are using freeswitch inside for a couple of things) I have tried simple port redirection with iptables, and while it SEEMS to work, the sip packets don't. so, does anyone know of a simple solution? I don't really want to replicate all the users, dial plans, sip trunk accounts on freeswitch. Just basically, anything coming in from a certain ip to the freeswitch port 5060, 'sipproxy' it to port 5080 on the internal system. then, anything sent out from the one certain internal ip, sourced on port 5080 to port 5060, send it to the external sip trunk provider. Using ip based authentication all around. I assume using an ip based condition with a FWD, (but the FWD would need to be generic) if call comes in from siptrunkprovider to 1000@freeswitch.secnap.com:5060 , I want it transparently forwarded to 1000@sipx.secnap.com:5080 if an internal call goes out to 5619995000@freeswitch.com:5060, I want it transparently forwarded to 5619995000@siptrunkprovider.com:5060 (yes, I tried simple port mapping. worked with tcp on http, and ALMOST worked with udp on tftp, but audio was mangled) sipx suggested purchasing an ingate siperaor. hard to justify $4 or $5K when I could replace sipx with a commercial system for that price. (yes, I could move everything to freeswitch, and I might, but I am under a deadline ) -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best in Email Security,2010: Network Products Guide * King of Spam Filters, SC Magazine 2008 ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________

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Lua script issue.
by jesse 26 Aug '10

26 Aug '10

I have a bridge.lua script: phone1 = argv[1]; phone2 = argv[2]; dialstring1 = "sofia/gateway/xyz.com/" .. phone1; dialstring2 = "sofia/gateway/xyz.com7" .. phone2; session1 = freeswitch.Session(dialstring1); session2 = freeswitch.Session(dialstring2, session1); freeswitch.bridge(session1, session2); freeswitch(a)xyz.com> luarun bridge.lua 16502222222 16503333333 2010-08-24 16:47:55.331681 [NOTICE] switch_channel.c:779 New Channel sofia/external/16502222222 [043e3bb2-afda-11df-b543-e5f35c280c99] +OK freeswitch(a)xyz.com> 2010-08-24 16:47:55.425462 [INFO] sofia.c:662 sofia/external/16502222222 Update Callee ID to "Caller Jesse" <43583> 2010-08-24 16:47:55.427493 [NOTICE] sofia_glue.c:3294 Pre-Answer sofia/external/16502222222! 2010-08-24 16:47:55.428468 [NOTICE] switch_channel.c:779 New Channel sofia/external/16503333333 [044cee96-afda-11df-b544-e5f35c280c99] 2010-08-24 16:47:56.360476 [INFO] sofia.c:662 sofia/external/16503333333 Update Callee ID to "16503333333" <16503333333> 2010-08-24 16:47:56.361475 [NOTICE] sofia_glue.c:3294 Pre-Answer sofia/external/16503333333! 2010-08-24 16:47:56.363477 [ERR] switch_cpp.cpp:1220 Channels not ready 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup sofia/external/16503333333 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] 2010-08-24 16:47:56.363477 [NOTICE] switch_cpp.cpp:972 Hangup sofia/external/16502222222 [CS_SOFT_EXECUTE] [NORMAL_CLEARING] 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session 25 (sofia/external/16503333333) Ended 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/16503333333 [CS_DESTROY] 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1220 Session 24 (sofia/external/16502222222) Ended 2010-08-24 16:47:56.364556 [NOTICE] switch_core_session.c:1222 Close Channel sofia/external/16502222222 [CS_DESTROY] As you can see the call gets dropped immediately after couple rings. however, it works well if i do like this: originate sofia/gateway/xyz.com/16502222222 &bridge(sofia/gateway/xyz.com/16503333333) What is the reason Lua script will fail? any difference between the two approaches? thanks! jesse

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Everyone come join IRC!
by Michael Collins 26 Aug '10

26 Aug '10

Hey all, We are having a big IRC day today so we'd love to have everyone join and help answer questions. Please join #freeswitch on irc.freenode.net. Thanks! -MC

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Sending fax problem
by pipiwei03 26 Aug '10

26 Aug '10

hello, I can receive and send a fax in php already. Now, I want to set a specific fxo port, not the whole gateway, to send a fax. What configuration should I set? The environment is described as follow: a freeswitch server based on winxp. server ip: 192.168.1.1 a WellGate 2644 <http://www.welltech.com/product_e_0r.htm> gateway which has 4 fxo and 4 fxs. each fxs/fxo port can register an sip account to freeswitch gateway ip: 192.168.1.2 i config my gateway as below in freeswitch: <gateways> <gateway name="faxgw"> <param name="username" value="0000"/> <param name="password" value="0000"/> <param name="realm" value="192.168.1.2"/> <param name="register" value="false"/> </gateway> </gateways> the gateway register 2 user as below <user id="pstn_fxo1"> <params> <param name="password" value="1234" /> </params> <variables> <variable name="user_context" value="default"/> <variable name="toll_allow" value="domestic,international,local" /> <variable name="effective_caller_id_name" value="pstn_fxo1" /> </variables> </user> <user id="pstn_fxo2"> <params> <param name="password" value="1234" /> </params> <variables> <variable name="user_context" value="default"/> <variable name="toll_allow" value="domestic,international,local" /> <variable name="effective_caller_id_name" value="pstn_fxo2" /> </variables> </user> in my php file, i can send the fax via fxo1 pstn line (use event_socket_request <http://wiki.freeswitch.org/wiki/PHP_Event_Socket>) and the command is "api originate sofia/gateway/faxgw/". $dest_phone_number. " &txfax(tmp.tif)" my problem is 1. How to send fax via fxo2 pstn line Should i bridge to user "pstn_fxo2" first and then dial number? I try to use "api originate sofia/internal/pstn_fxo2 ".$dest_phone_number. " &txfax(tmp.tif)" but it didnt work 2. In <gateway> config The "username" and "password" seem no effect. I just set the "realm" to my gateway ip and it works. And the "register" i dont know it too. The document didnt describe it clearly, can u explain it for me Thank you sincerely. Joy

1 0

How could install PHP ESL?
by jesse 26 Aug '10

26 Aug '10

The current FS version doesn't include source code for PHP. What are the steps to install it on top of my current version? where to get the source? how to build and install? Iif you guys want to keep the new system slim by get rid of PHP, please at least keep a doc about how to add it in case of need. thanks! -jesse

7 13

Still can't dial gateway from ZAP phone.
by Jim 26 Aug '10

26 Aug '10

Michael Collins gave me a suggestions a while back: I looked in mod_openzap.c and I didn't see any references to channel variables. However, you have context and dialplan options. I suggest that you create a dialplan context just for your FXS port(s). Try this. Create conf/dialplan/fxs-ports.xml: <include> <context name="fxs-ports"> <extension name="outbound"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="toll_allow=local,domestic,international"/> <action application="transfer" data="$1 XML default"/> </condition> </extension> </context> </include> Then in your openzap.conf.xml change the context for the analog span(s) with the FXS ports: <param name="context" value="default"/> Restart FS after making these changes and then give it a shot. You should see the call from the analog phone going into context "fxs-ports" and then get transferred over to the default context where it will act like your SIP phones because we manually set the ${toll_allow} chan var. -MC Unfortunately that did not work. The "default_gateway" variable used by this line: Dialplan: OpenZAP/1:1/17705550068 Action bridge(sofia/gateway/${default_gateway}/17705550068) ended up looking like: EXECUTE OpenZAP/1:1/17705550068 bridge(sofia/gateway//17705550068) whereas a successful dialout looks like: EXECUTE sofia/internal/1002(a)192.168.2.51 bridge(sofia/gateway/gw4.telasip.com/17705550068) Somehow the information in the directory/default/default.xml file never got included and I'm not sure how to fix it. Thanks for any guidance. Jim.

3 4

Inbound and then outbound call?
by Malay Thakershi 26 Aug '10

26 Aug '10

Hello, It would be a great help if someone can guide me. 1. I would like to first receive a call, perform certain validations. (Able to do this via mod_managed application that handles call from dialplan). 2. Now, I would like to dial out to a PSTN number so that received call is connected to this new outbound number. How can this be done? Do I use Originate from within my .NET (mod_managed) code? Do I get charged for both incoming and outbound call until the entire session ends? Is there a way to receive call, validate and then sort of transfer and then terminate the received call so I do not get charged for both? Please help. Thank you.

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Platform requrest for the Jira issue management system
by Vincent Stemen 26 Aug '10

26 Aug '10

Hopefully this is an approprate place for this request. I would like to see "DragonFlyBSD/GCC" added as a Platform option. Regards, - Vince

3 6

MWI doubt
by François Delawarde 26 Aug '10

26 Aug '10

Hello, I wish to have a single domain across multiple profiles. For registrations and calls it works perfectly. Having a shared database for all profiles, using sofia_contact with "<anyprofile>/<user>@<domain>" gives back the full dialstring with the profile where the phone is currently registered. Now I was trying to do some MWI accross those profiles, and it appears that some phones have the NOTIFY message every time they register, and some other do not receive. Any reason? Also, when a phone is configured to SUBSCRIBE for MWI, it doesn't receive NOTIFYs when a new message is on the voicemail. The only thing that appears wrong is this console message: 2010-08-25 16:09:57.578144 [ERR] sofia_presence.c:316 Cannot find profile <domain> - Does the profile need to have an alias to a particular domain for presence/MWI subscriptions to work? - Is it possible somehow to have multiple profiles aliased to a same domain? If not, what is the recommended way to share presence or MWI across profiles? Thank you, François.

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