November 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Install ASTPP
by Joao Cesar / Consultor TI 05 Jan '13

05 Jan '13

Em 25/10/2012 10:46, Michael Jerris escreveu: > Try sending this email to the mailing list? > > On Oct 25, 2012, at 8:41 AM, Joao Cesar / Consultor TI <consultoriaticesar(a)gmail.com> wrote: > >> Anyone know how to install the ASTPP really, because the tutorial site, have not the slightest condition to do. >> Thank you in advance. >> >> -- >> Joao Cesar S Pereira >> Consultor TI >> ---------------------------------------- >> (31)3044-2576 >> (31)8626-6943 >> ---------------------------------------- >> Gentileza, se nao houver necessidade >> não imprima. A natureza agradece. >> ---------------------------------------- >> >> -- Joao Cesar S Pereira Consultor TI ---------------------------------------- (31)3044-2576 (31)8626-6943 ---------------------------------------- Gentileza, se nao houver necessidade não imprima. A natureza agradece. ----------------------------------------

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Sofia fails to detect offline gateway
by Steven Ayre 03 Jan '13

03 Jan '13

Hi Guys, A couple of our gateways have been shut down for hardware upgrades. However, the FreeSWITCH servers didn't spot the gateways go offline so they're still trying to send calls to them. I see this in the logs: 2011-01-11 09:35:05.396610 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:05.396610 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:13.408674 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:13.408674 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:21.420884 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:21.420884 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:29.458114 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:29.458114 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:37.472424 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:37.472424 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:46.475934 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:46.475934 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:55.489967 [WARNING] sofia.c:3981 Ping failed gw4 with code 503 - count 1/0/10, state UP 2011-01-11 09:35:55.489967 [WARNING] sofia.c:3981 Ping failed gw3 with code 503 - count 1/0/10, state UP ... you get the idea. I've done some testing and it happens both when FS is running and the gateway goes offline (code 408 then) and when FS starts up with the gateway already offline (code 503 as above). Both IP addresses are entirely offline. I had expected these would go to state DOWN but they haven't, I just get the above repeating. The gateway definitions look like this: <gateway name="gw3"> <param name="realm" value="a.b.c.d"/> <param name="username" value="nouser"/> <param name="password" value="nopass"/> <param name="register" value="false"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="5"/> <param name="ping-max" value="10"/> <param name="ping-min" value="1"/> <param name="retry-seconds" value="5"/> <param name="expire-seconds" value="60"/> </gateway> Is there a mistake in my configuration? -Steve

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mod_xml_curl - missing dialplan variable (domain of originating user for blind transfer)
by Cal Leeming [Simplicity Media Ltd] 31 Dec '12

31 Dec '12

Hi guys, In a nut shell, it appears that when attempting to perform a blind transfer under certain conditions (explained below), mod_xml_curl does not expose the originating domain in a clean format. My initial plan was to find the point where these variable were being generated, look at what was available, then add an extra variable for the domain and submit a patch. Sadly my C isn't great and I hit a brick wall, so if anyone can help out, I will ensure the mod_xml_curl documentation is updated and/or assist with any patching/testing required. Please take the following scenario; * Extension 2000 calls an external number via a gateway (i.e. bridge sofia/gateway/name/e164_number_here). * Call connects fine, audio stays good, no disconnection problems etc. * Call is blind transferred to another extension As a result, the following is determined; * User initiating the blind transfer is 2000 * Domain initiating the blind transfer is c1881.voiceflare.co.uk * Destination number of the call is 447866123456 * Number to blind transfer to is 2001 * Call to mod_xml_curl is made It makes reference to the User in the following 'clean' variables (by clean, I mean variables that just contain 2000 and don't require mangling to extract the info); u'Caller-ANI': u'2000', u'Caller-Username': u'2000', u'Caller-Caller-ID-Number': u'2000', u'Hunt-ANI': u'2000', u'Hunt-Caller-ID-Number': u'2000', u'Hunt-Username': u'2000', u'variable_last_sent_callee_id_number': u'2000', u'variable_sip_from_user': u'2000', It also has the User in the following unclean variables; u'variable_bridge_channel': u'sofia/external/ 2000@c1881.voiceflare.co.uk:5060', u'variable_sip_from_uri': u'2000(a)89.238.182.137', u'variable_sip_full_from': u'"foxx" <sip:2000@89.238.182.137 >;tag=XryjFQp3rB2NF', u'variable_sip_h_Referred-By': u'"foxx" < sip:2000@c1881.voiceflare.co.uk:5060>', However, it only references the domain in the following unclean variables; u'variable_bridge_channel': u'sofia/external/ 2000@c1881.voiceflare.co.uk:5060', u'variable_sip_h_Referred-By': u'"foxx" < sip:2000@c1881.voiceflare.co.uk:5060>', u'variable_sip_refer_to': u'<sip:2001@c1881.voiceflare.co.uk:5060>', Lets say that we want to determine the user/domain that has initiated this transfer, doing so would mean mangling with one of those above variables, which seems a bit dirty (plus it is not clear which is the correct one to use). I have attached the SIP trace of the entire blind transfer event, and the full mod_xml_curl request being sent. If anyone could answer the following, it'd be much appreciated; * Should there be an improvement patch that exposes the domain of the user that originated the blind transfer? * Are there better/alternative ways to extracting the domain of the user that originated the blind transfer? Many thanks Cal

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FreeSwitch Video Calls Bria iPhone Client
by Mike Burlingame 27 Dec '12

27 Dec '12

I have been playing around trying to get my iPad and iPhone using Bria to work with video it *seems* that FreeSwitch is passing the signaling to both devices and FreeSwitch is holding the media due to the NAT issue of the devices - I just wanted to know before I spent too much time on this (not a high priority) if anyone else has gotten FreeSwitch to work with Bria for iPhone or Android? Thanks

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Cisco-signed SSL Certificates/Alternative phone suggestions
by Andrew Cassidy 21 Dec '12

21 Dec '12

Hi guys, just looking for a little help. I've been using Cisco SPA50x phones for a little while now, they have some pretty nice features for the price, including the ability to write custom applications in XML. However, to run them over HTTPS instead of HTTP, you need a certificate signed by Cisco, which I'm having difficulty obtaining. So my questions are: 1. Can anyone help me obtain such a certificate? I have the CSR already to go. 2. Are there any similarly priced and featured phones available that you guys have experience with? Thanks guys, -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T <info(a)cassidywebservices.co.uk> *03300 100 960 *F<info(a)cassidywebservices.co.uk> *03300 100 961 *E <info(a)cassidywebservices.co.uk> *andrew(a)cassidywebservices.co.uk *W <info(a)cassidywebservices.co.uk> *www.cassidywebservices.co.uk

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Mod_opal & PTLIB
by Phil Quesinberry 12 Dec '12

12 Dec '12

Im trying to compile mod_opal here but I cant get it to work. Ive tried the buildopal.sh script, the manual install, sacrificial offerings, etc. to no avail. I was getting an error about PTLib being too old, so I grabbed PTLib 2.10.7 source here: http://www.linuxfromscratch.org/blfs/view/svn/general/ptlib.html Trying to compile it, I get a No operating system selected error. Ive pasted in the make attempt below, Distro is CentOS 5.8. It looks like a lot of other folks have had/are having problems with mod_opal and ptlib, but Im not seeing much in the way of solutions. Has anyone else gotten past this problem? Any pointers? (no pun intended) [root@Tyrion ptlib]# make Setting default PTLIBDIR to /root/ptlib make[1]: Entering directory `/root/ptlib' make[2]: Entering directory `/root/ptlib' make[3]: Entering directory `/root/ptlib' make[3]: Leaving directory `/root/ptlib' make[2]: Leaving directory `/root/ptlib' set -e; if test -d /root/ptlib/src ; then make -C /root/ptlib/src optshared; fi; if test -d /root/ptlib/plugins ; then make -C/root/ptlib/plugins optshared; fi; make[2]: Entering directory `/root/ptlib/src' make[3]: Entering directory `/root/ptlib/src' [CC] ptclib/psasl.cxx In file included from /root/ptlib/include/ptlib/object.h:44, from /root/ptlib/include/ptlib.h:47, from ptclib/psasl.cxx:35: /root/ptlib/include/ptlib/unix/ptlib/platform.h:555:2: error: #error No operating system selected. /root/ptlib/include/ptlib/mutex.h:109: error: âPThreadIdentifierâ does not name a type /root/ptlib/include/ptlib/thread.h:292: error: âPThreadIdentifierâ does not name a type /root/ptlib/include/ptlib/thread.h:293: error: âPThreadIdentifierâ does not name a type /root/ptlib/include/ptlib/thread.h:403: error: âPThreadIdentifierâ does not name a type /root/ptlib/include/ptlib/syncthrd.h:326: error: âPThreadIdentifierâ was not declared in this scope /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 1 is invalid /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 3 is invalid /root/ptlib/include/ptlib/syncthrd.h:326: error: template argument 4 is invalid /root/ptlib/include/ptclib/cypher.h:382: error: âPUInt32lâ does not name a type make[3]: *** [/root/ptlib/lib_linux_x86_64/obj/psasl.o] Error 1 make[3]: Leaving directory `/root/ptlib/src' make[2]: *** [optshared] Error 2 make[2]: Leaving directory `/root/ptlib/src' make[1]: *** [optshared] Error 2 make[1]: Leaving directory `/root/ptlib' make: *** [default] Error 2 Many thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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Avaya SIP Phones?
by Sean Devoy 10 Dec '12

10 Dec '12

Hi, We have a potential customer asking us to give them a bid with AVAYA phones!!! Does AVAYA have any true SIP phones? Do they work with FreeSwitch? Thanks, Sean

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FreeTDM ftmod pri tap error
by Gopalakrishnan N 10 Dec '12

10 Dec '12

Hi, Am trying to install ftmod pri tap by following this link http://wiki.freeswitch.org/wiki/FreeTDM#Tapping and enabled --with-pritap while doing ./configure, and while doing make I get error. Can someone help me on this. Do I need to update any patch for tap? My installation log is attached here. Regards. Gopal.

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Domains and profiles
by Lawrence Conroy 06 Dec '12

06 Dec '12

Hi Folks, I've started a new thread as it's not quite the same issue, and domains & profiles have confused the heck out of me every time I have developed a new setup for fS. I have sometimes had to hack/hard-doce the dialstring to make multiple domains in one profile work, had hours of fun with presence, db and force register settings, and have still had some odd gotchas that have required extensive meditation. [... and yes, I have read the 1.0.6 bridge book; I'm trying to abstract these elements ] Coming at this from standards/specs and rolling my own SIP stacks, sofia/fS seems to use the term "domain" differently from sipdomain, and alias seems to be tied to the directory (and thus to the profile listed in a directory file), but I'm not sure. so ... Before I capture to the sofia conf xml wiki page, I have a couple of questions on the sip-profile XML setup; Q: Is there a particular reason why there's a parameter called alias and an (entirely different) setting also called alias? The sofia conf xml wiki's comment on the setting "alias" shows I'm not alone. I agree that's what it appears to be doing, but can we nail this down please (and what happens if an external client uses this connection to register and call)? In the current sofia conf xml wiki page, the domain setting is not exactly well documented :). The current internal.xml vanilla example from git (as of time of writing) has the following lines: ------------------------- ...   <aliases>  </aliases>  <gateways> <X-PRE-PROCESS cmd="include" data="internal/*.xml"/> </gateways> <domains>     <domain name="all" alias="true" parse="false"/> </domains> ... ------------------------- This stuff is entirely missing from the sofia.conf.xml wiki page, and it IS really important. Q: what's the default value for the alias parameter in the domain element? -- it is missing from the first example. Q: if there is more than one profile, what's the impact of setting parse = true in one (or all) of the profiles' XML files? (or parse = false, or missing the parameter altogether)? AFAICT, the gateways get pulled in via the pre-process directive just fine, regardless of the value of the parse parameter -- it works for me, at least. Q: if there is more than one profile, what's the impact of putting domain name="all" into one (or all) of the profiles' XML files? Ideally, having more than one sipdomain tied to one profile "would be good", but aliases doesn't do that -- as the git file says, these are aliases for the profile name. Before I start scribbling, Answers on a postcard to this ML, please. all the best, Lawrence

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Call disconnect upon re-invite - FS tears down call
by Phil Quesinberry 05 Dec '12

05 Dec '12

I'm experiencing an odd issue here where a direct media call (bypass media mode) between two Aastra endpoints is disconnected when one of the endpoints sends a re-invite 2 minutes into the call, presumably to refresh the session - RTP ports & codecs remain unchanged. FreeSWITCH passes the invite on to the other endpoint which replies with its own invite, again keeping the same call parameters, ports/codecs etc. FS then sends a 491 - Request Pending to both endpoints, acknowledges the second endpoint's invite and tears down the call. This seems wrong. is this expected behavior? I went ahead and filed a Jira on this just in case - it includes all trace info: http://jira.freeswitch.org/browse/FS-4754 Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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