- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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Garbled audio on playback
by Young, Christopher 25 Jun '13

25 Jun '13

Hi all, First time poster here and relatively new to FreeSWITCH so please forgive me if my terminology is not quite right. I've encountered a strange problem with audio playback. If I use either switch_ivr_play_file() or switch_core_session_execute_application() with the playback application to play an audio file to a session then I get garbled output and lots of, "Deactivating write resampler" messages in the log. However, if I use switch_ivr_broadcast() (or the uuid_broadcast command) then it plays fine (and there is only a single occurrence of the resampler message). I'm a bit confused because if I understand things correctly then all of the methods mentioned above ultimately go through switch_ivr_play_file() anyway so why would some work and some not? The problem seems similar to an item I found in JIRA (FS-5236) but I tried an updated version which includes that fix and it didn't make any difference. Any help would be appreciated. Regards, Chris Chris Young Software Engineer direct: +44 (0) 118 943 9249 web: www.enghouseinteractive.com<http://www.enghouseinteractive.com/> | email: cyoung(a)enghouse.com<mailto:cyoung@enghouse.com> A Division of Enghouse Systems Ltd.<http://www.enghouse.com/> (TSX: ESL)

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Codec selection in reinvite to a-leg in bypass_media_after_bridge
by Javier Gallart 25 Jun '13

25 Jun '13

Hello list, let me describe the call scenario: We have an established call with media traversing Freeswitch. In some cases we'd like to get FS out of the media path, and for those cases we set the variable bypass_media_after_bridge to true. It works fine, the reinvite is sent in both directions with a new c= address. However in some cases this can result in no audio calls. Imagina A sends the call to profile internal with inbound_codec preference: PCMU,G729,G723 A->FS offer: PCMU,G729 FS->B offer: PCMU,G729 B->FS answer: G729 FS->A answer: G729 The reinvite sent to the B-leg contains a single codec (G729); however the reinvite to a-leg contains all the codec defined in the profile. If A happens to OK to the reinvite with PCMU we will have a no-audio call . I've tried several things but I haven't been able to find if this is actually feasible. Has anyone tried the same? Thanks Javi

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Fax T38 and disable_udptl_auto_adjust not working
by Martin Soerensen 25 Jun '13

25 Jun '13

Hi all. Hope that some on can help...... I have a problem with t38 from at audiocodes gateway to FS as fax receiver.... FS changes the UDP port form 15000 to 15002. I have tried setting <param name="disable-rtp-auto-adjust" value="true"/> in <gateway> and in the profile settings section without any luck I have also tried setting the <action application="set" data="disable_rtp_auto_adjust=true"/> in the dial plan but still it's not working. My audiocodes GW expects to get an answer on the same port that it's sending on Please HELP :o) Regards Martin

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help on mod_callcenter
by Ashish gautam 25 Jun '13

25 Jun '13

I want to get events on a script which is listening on a particular port on my system from mod_callcenter. The events are not being sent to this script. Also when the script is listening on that port, even the agents are not being rung and Only the CHANNEL_HANGUP event is being sent. -- Ashish

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Re: [Freeswitch-users] 1.4beta - rtmp registration doesn't work with domain name
by Matthew Cordes 25 Jun '13

25 Jun '13

Hi William, I think it's a false alarm. I'm experimenting with mod_xml_curl's 'a1-hash' for directory lookups and from what I can tell mod_rtmp doesn't support this. It's chocking at rtmp.c:229 with the message 'Authentication failed for XXX empty password not allowed'. This seems to be the root of my problem. Sorry for the confusion and it's unfortunate rtmp doesn't look like it's going to play nicely (unless I'm missing something). -Matt ---------- Forwarded message ---------- > From: William King <william.king(a)quentustech.com> > To: freeswitch-users(a)lists.freeswitch.org > Cc: > Date: Mon, 24 Jun 2013 17:23:30 -0700 > Subject: Re: [Freeswitch-users] 1.4beta - rtmp registration doesn't work > with domain name > Can you run a git bisect to see where this feature breaks for you? > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king(a)quentustech.com > > On 06/24/2013 07:10 AM, Matthew Cordes wrote: > > Hello, > > > > Registering via the rtmp endpoint using a domain name for the domain (as > > opposed to an ip address) doesn't seem to be working in the 1.4beta > branch. > > > > I tried setting the domain in config/vars.xml to both the ip address and > > the domain name without luck. Either works with rtmp on the 1.2 branch > > on the same machine with similar configuration. > > > > For comparison, registering from a sip softphone (not using rtmp) works > > fine using either the domain name or ip address in both 1.2 and 1.4. > > > > Any thoughts? Is this a deliberate change in 1.4? > > > > -Matt >

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Is there a mod_xml_curl password hashing scheme that doesn't include the domain?
by Matthew Cordes 25 Jun '13

25 Jun '13

Hi, I'm interested in using mod_xml_curl's 'a1-hash' hashing to avoid storing plain text passwords. Unfortunately I'm finding building that hash cumbersome because it includes the domain. I don't see anything in the documentation, but is there an option to to provide a md5 hash of just the password when doing a directory lookup via mod_xml_curl? Or perhaps some way to tell freeswitch to ignore the domain (or use some other value) when generating the hash? -Matt

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1.4beta - rtmp registration doesn't work with domain name
by Matthew Cordes 25 Jun '13

25 Jun '13

Hello, Registering via the rtmp endpoint using a domain name for the domain (as opposed to an ip address) doesn't seem to be working in the 1.4beta branch. I tried setting the domain in config/vars.xml to both the ip address and the domain name without luck. Either works with rtmp on the 1.2 branch on the same machine with similar configuration. For comparison, registering from a sip softphone (not using rtmp) works fine using either the domain name or ip address in both 1.2 and 1.4. Any thoughts? Is this a deliberate change in 1.4? -Matt

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voicemail_say_phone_number
by Yuriy Nasida 24 Jun '13

24 Jun '13

Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise.Thanks.

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