August 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Speak application w/ SSML over IES
by Ben Langfeld 03 Aug '12

03 Aug '12

Attempting the following via IES results in flite reading out the XML document, rather than rendering it as SSML, regardless of what escaping I try. Can anyone clarify how this is supposed to be done? SendMsg 0e66ee48-ddb6-11e1-a46e-b7cac2dab4cc call-command: execute execute-app-name: speak execute-app-arg: %[punchblock_component_id=e8f7ebcb-d107-4d6e-bf9f-fca4d7b50d01]flite|kal|<speak xmlns="http://www.w3.org/2001/10/synthesis" version="1.0" xml:lang="en-US">Hello Fernando</speak> Regards, Ben Langfeld

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SOLVED: Asterisk don't like multiple m= lines in the SDP [Was: Error: 88 Incompatible Destination after upgrade]
by A E G 03 Aug '12

03 Aug '12

Well turns out learning to use Google better always helps. Found a nugget of wisdom that <action application="set" data="sdp_m_per_ptime=false"/> should've actually been in the vars.xml Seems to have fixed it. On Fri, Aug 3, 2012 at 12:01 AM, A E G <all.eforums(a)gmail.com> wrote: > Gents, > > running version: 1.2.0-rc2+git~20120731T213556Z~e97da8e20a > > have been fighting with this for an hour or more...have done a bit of > research on the list and Google itself, scoured the Wiki etc. but can't > seem to figure out where to set the codecs to be "well recd" by the remote > SIP peer. This used to work fine until I upgraded 2 days ago and it stopped > working without my doing / changing anything. > > The remote SIP peer is {*} which doesn't like multiple m= lines with > differing ptime values. > > Debug on near-end says: "Originate Resulted in Error Cause: 88 > [INCOMPATIBLE_DESTINATION]" > > Full SDP here: > > v=0 > o=FreeSWITCH 1343934933 1343934934 IN IP4 192.168.1.80 > s=FreeSWITCH > c=IN IP4 192.168.1.80 > t=0 0 > m=audio 28884 RTP/AVP 98 8 3 101 13 > a=rtpmap:98 L16/16000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > m=audio 28884 RTP/AVP 0 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:40 > a=sendrecv > > Far end says: Rejecting non-primary audio stream: audio 28884 RTP/AVP 0 > 101 13 > > I have tried to play around with the codec globals in vars.xml, to no > avail. > > have also added stuff directly in the dialplan like so: > > <action application="set" data="hangup_after_bridge=true"/> > <action application="set" data="absolute_codec_string=PCMU@40i"/> > <action application="set" data="sdp_m_per_ptime=false"/> > <action application="bridge" data="sofia/gateway/mycore1/1111"/> > > > but no dice. > > Tried to remove the "absolute_codec_string", still no dice. > > I read about the "sdp_m_per_ptime" that lets me "cheat" but that ain't > doing anything either. > > What gives? > > Thx in advance > > > > > >

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SIMPLE msg between two freeswitch servers
by Carlos Flor 03 Aug '12

03 Aug '12

I am trying to send a SIP SIMPLE message from handset 101 registered to switch A to handset 202 registered on switch C. Switch B is in the middle and is a gateway for switch A, so it looks something like this: 101 ----- FS A -----gw------- FS B -----------gw------- FS C --------- 202 I was able to get something working with the chatplan by setting the "to" parameter to point to FS B and then on FS B setting the to parameter to point to FS C. I don't know if this is even the best way to do it, but that's how I got it to work. What I would like, however, is to use TLS on all links. I currently have TLS on all links for phone calls, but I guess the chatplan doesn't use the same logic as the dialplan so by default, it is using UDP between the switches. I have been looking through the source trying to find out what I can set to enable TLS for SIMPLE between switches but I have been unsuccessful thus far. Any help would be greatly appreciated. Thanks!

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G711-16?
by Phil Quesinberry 03 Aug '12

03 Aug '12

Does FS support G711/PCMU 16K? If so, how do you specify it in the codec list? I can't seem to find any info on it and 'show codecs' doesn't show it so my first assumption is that it doesn't. Since Aastras and other endpoints support it, I thought I'd better ask before assuming that was the case. Thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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Directory xml_curl, gets Can't find user
by ServTelar 03 Aug '12

03 Aug '12

Hi all It seems this is a very old one, but certainly I didn't find any solution to this. Using XML for directory everything works fine. But if I try using xml_curl, sending response: <document type="freeswitch/xml"> <section description="FreeSWITCH Directory" name="directory"> <domain name="10.8.0.70"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_user}@${dialed_domain})}"/> </params> <groups> <group name="default"> <users> <user id="2692128"> <params> <param name="a1-hash" value="7d73f412ca332fedd81a297b64c0b271"/> <param name="vm-password" value="2128"/> </params> <variables> <variable name="accountcode" value="2692128"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="2692128"/> <variable name="effective_caller_id_number" value="2692128"/> <variable name="outbound_caller_id_name" value="2692128"/> <variable name="outbound_caller_id_number" value="2692128"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> Registration is accepted: Registrations: ================================================================================================= Call-ID: 939540755 User: 2692128(a)10.8.0.70 Contact: 2692128 <sip:2692128@10.0.0.10:5060;line=3dfb73927e26966;fs_nat=yes;fs_path=sip%3A2692128%4010.94.56.6%3A5060%3Bline%3D3dfb73927e26966> Agent: iSoftPhone 3.4110 Status: Registered(UDP-NAT)(unknown) EXP(2012-08-03 11:35:30) EXPSECS(1688) Host: demo IP: 10.94.56.6 Port: 5060 Auth-User: 2692128 Auth-Realm: 10.8.0.70 MWI-Account: 2692128(a)10.8.0.70 But if I made a call pointing to that user I just got: 2012-08-03 11:05:38.261385 [WARNING] mod_dptools.c:3294 Can't find user [2692128(a)10.8.0.70] 2012-08-03 11:05:38.281377 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2012-08-03 11:05:38.281377 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT] Domain are set both to the IP: <X-PRE-PROCESS cmd="set" data="domain=10.8.0.70"/> <X-PRE-PROCESS cmd="set" data="domain_name=10.8.0.70"/> And matches with the internal freeswitch@internal> sofia status Name Type Data State ================================================================================================= 10.8.0.70 alias internal ALIASED internal profile sip:mod_sofia@10.8.0.70:5060 RUNNING (0) Can someone drop me a line to point where I should look to? Thanks in advance Gustavo

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Help so inbound-late-negotiation=true and SRTP can coexist in same profile?
by Robert Hadley 03 Aug '12

03 Aug '12

Hi FS users, We want to use <param name="inbound-late-negotiation" value="true"/> in the internal sip profile, and <action application="inherit_codec=true"/> in the dialplan so as to minimize codec transcoding in Freeswitch. But we also have other devices registered to the same sip profile using TLS/SRTP. With this combination of settings SRTP is broken on the B-leg, resulting in RTP media. Is there a way to override the sip profile parameter inbound-late-negotiation on a per call/channel basis? Freeswitch's default internal sip_profile looks like it should support both UDP (5060) and TLS/SRTP (5061), but the above limitation makes the internal sip profile unusable for TLS/SRTP. Is there any way for SRTP and late negotiation to coexist in the same profile? Thanks, Robert

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How to file FreeSwitch Enhancements requests?
by Mario G 03 Aug '12

03 Aug '12

Is there an official way to suggest FreeSwitch enhancements? Where should it go and how much doc/explanation is needed? In this case: 1. in bridge: Add the ability to specify variables that apply only to one group_call but all users in that group. 2. In bridge: Add the ability to allow leg_delay_start, etc. to work using enterprise syntax. I posted this yesterday which may explain more: (has gotten no response) What I need: to have different variable/settings for each target group in a bridge command, and allow leg_delay_start to work. It's related to trying to fix other issues and holding up progress for them. Thanks for any help, Mario G I scoured the wiki and tried many things. Normally, enterprise syntax would solve this, but there is a catch: you can't use leg_delay_start which I need, it is ignored when using enterprise. I found that using brackets [] only applies the variable to the first user in the group, not the whole group so that won't work. What I have now but need to have alert_info moved/apply to the mgt group only: <action application="bridge" data="{originate_timeout=45,alert_info=n=${lua_ringtone}}${group_call(mgtbria@${domain_name}+A)},${group_call(mgt@${domain_name}+A)},[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551212,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551313"/> This solves variables for groups but breaks leg_delay_start, etc.: <action application="bridge" data="<originate_timeout=45>}${group_call(mgtbria@${domain_name}+A)}:_:{alert_info=n=${lua_ringtone}${group_call(mgt@${domain_name}+A)}:_:[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551212,[leg_delay_start=20,leg_timeout=23]sofia/gateway/${dial_gateway}/19165551313"/>

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Anyine have Bria iOS tipes for FreeSwitch?
by Mario G 03 Aug '12

03 Aug '12

I am at my wits end after several months of Bria problems. Counterpath keeps pointing the finger at FreeSwitch. I am wondering if anyone here has Bria working well and if they can share tips user definitions? Bria is set for TCP to reduce battery usage. Thanks. Mario G

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External profile & NAT
by ocset 03 Aug '12

03 Aug '12

Hi I would like to allow a person with a softphone app to register with Freeswitch (extension 1111) when they are outside the office network. I would like to to implement this using an external profile. According to the wiki page (http://wiki.freeswitch.org/wiki/External_profile) I just need to create a copy of the "conf/sip_profiles/external.xml" file and change the port number to 5090 (and also open the port on the modem). When I do a "sofia status", this new gateway does not show up and to be honest, I don't quite understand how that works and what else to read to make sense of what to do. What are the ways in which other Freeswitch admins are granting access to external devices? ps. just to confirm, do I set external_rtp_ip and external_sip_ip to the fix IP address that my ISP has given me - the one that is shown by www.whatsmyip.org? Please help Thanks O.

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ESL Log to console
by David Villasmil 03 Aug '12

03 Aug '12

Hello Guys, I'm starting off with ESL, which is cool, but I'm trying to log to the console like Lua's "freeswitch.consoleLog("info", "BLAH BLAH");" doing it like: my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); $con->execute("log", "1, BlahBlah"); But nothing gets in the log files or console... and I can't find any documentation as to how to log using "execute"... any ideas? Thanks! David

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