November 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_ldap
by Daniel Browne 23 Nov '09

23 Nov '09

I'm new to Freeswitch and I'm looking for some help using mod_ldap to authenticate SIP endpoints to an LDAP database on registration. I have successfully installed the Freeswitch trunk and OpenLDAP 2.4.18 and I have compiled mod_ldap. I have setup a config file for mod_ldap. It is active and I can see its config in the compiled Freeswitch config file. I have turned on SIP traces, sofia logging and console logging in the Freeswitch command line. However, when I register a SIP phone to my Freeswitch server, I see no reference to mod_ldap in any logs. I have not set up the correct schema in ldap yet, so I would expect to see some indication that the module has searched my ldap server and found no useful information. My phone registers correctly and the normal tests (eg. music on hold) work. I am not sure if mod_ldap falls back on the usual xml config files if it fails to find information on the specified ldap server (as mod_xml_curl does), so I can't be sure if it is working or if my phone is simply registering in the normal way. Can anyone give me pointers on where to look next? If I can just get some feedback from the module I should be able to work out what to do. Thanks -- Grimsqueaker "Even a fool, when he holdeth his peace, is counted wise." Proverbs 17:28a

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How to pick up someone's phone remotely.
by Piotr Żurek 23 Nov '09

23 Nov '09

Hello. Thank You developers for Freeswitch. I have installed it lately and it's working quite nicely, but I have one problem: I need to mimic behavior of my current analogue PBX installation using Freeswitch. This is the scenario: In the office with a few desks (extensions 1000-1010) and only one person behind one of desks (whatever extension - in example 1000). 1. There's incoming call on _one_ of extensions 1001-1010 2. The person on extension 1000 wants to answer this call on his phone so dials #37 and this call is redirected to his phone. That's how it works on my office on analogue PBX system. Anyone can answer a call from any other phone as long as it hasn't been answered already. I tried to use the intercept action (with global example in default config) but it's not what I need because it intercepts the call even if it's already answered. I need to intercept all but only unanswered calls. I tried to use Redirect but it does not work on other's extensions call's (or does it?). Please help. Peter Żurek

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Clarification about channel variables please.
by Lon Baker 23 Nov '09

23 Nov '09

Are either global or regular channel variable mutable during a call? Or can they only be set before and after? Any clarification would help, since the existing wiki doesn't make it clear. Lon

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Call quality problem w/ Snom M3's
by Joseph L. Casale 22 Nov '09

22 Nov '09

Not sure where to start with this one, the outgoing leg to our sip provider sounds perfectly fine but with the our M3's the incoming leg is super choppy. Using twinkle on my laptop yields good results in both directions so there must be an issue with just the snoms, their firmware was very old but updating it didn't help. Our sip provider uses ulaw and the snoms and vars.xml both have ulaw set as top pref. Any ideas what to look at next? Thanks, jlc

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Broadvoice 32 transcoding support?
by Gaurav Singh 22 Nov '09

22 Nov '09

Hi, Does freeswitch support transcoding between broadvoice (BV32 ) and G711 ? Did anyone try using freeswitch with Xten/counterpath Sip phone configured with broadvoice32? Also, can someone recomend another free sip phone supporting BV 32? Thanks Gaurav

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IP1001 Setup
by David V. Fansler 22 Nov '09

22 Nov '09

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Analog phone with ATA. Phone Won't Dial Out, but can receive calls
by Dave Stevenson 22 Nov '09

22 Nov '09

Hi, I have just purchased an ATA (Pluscom SIP VoIP ATA, model VPA-11) to try to use a normal (analogue) cordless phone with FreeSwitch. I have got the ATA setup and talking to FreeSwitch, it has registered the right extension and can receive and pick-up incoming calls. However, I can't dial numbers successfully, I can get the dial tone and dial the numbers, but the target numbers (internal or external) are not recognised by FreeSwitch. I have pasted a dump in the pastebin, but I can't see anything that tells me what the problem might be. Can anyone suggest what the problem might be please ? Regards Dave

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RTP issues (possibly nat-related)
by Jason White 21 Nov '09

21 Nov '09

I have upgraded FreeSWITCH several times recently for testing purposes. Also, my router's configuration has changed slightly as I have moved from tunneled IPv6 to a new native IPv6-over-ADSL trial. However, the problem now is related to my ISP's IPv4-only SIP service, and the symptoms are as follows. 1. If I call a test number, sometimes it all works perfectly. 2. On other occasions (with no discernible pattern) the call connects but no audio is received from the remote end. When this occurs, tshark shows that rtp packets are being sent out to the correct IPv4 address of the server. I am using Stun to handle nat, as my router does not support any of the nat configuration protocols. I want to establish whether it's a router issue or a FreeSWITCH problem. The router is going to be replaced eventually with a small form-factor Linux box and an ADSL2+ card from Traverse Technologies (http://www.traverse.com.au/) but given my priorities at the moment, it won't happen until next year. I can compare SIP traces of that would help.

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change event value
by Eli Hayun 21 Nov '09

21 Nov '09

Hi Is there is a way to intercept an event (for example : REGISTER) and change one of its parameters (for example: the extension number) and fire up the corrected event? I need it to set the speedial of the phone value to be "**xxxxx" but to make it register as "xxxxx" Thanks Eli

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freeswitch.spec patch
by Igor Neves 21 Nov '09

21 Nov '09

Hi, Attached it's a patch that corrects the problem when doing upgrade to other older version of freeswitch rpm the freeswitch user was being deleted. This patch was made against freeswitch.spec from freeswitch 1.0.4. It should be applied with "patch -p0 < freeswitch.spec". Cheers, -- Igor Neves<igor.neves(a)3gnt.net> 3GNTW - Tecnologias de Informação, Lda SIP: igor(a)3gnt.net JID: igor(a)3gnt.net ICQ: 249075444 MSN: igor(a)3gnt.net TLM: 00351914503611 PSTN: 00351252377120

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