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Bridge two calls
by Ken Fulmer 30 Mar '10

30 Mar '10

I need to bridge two calls together: one from an ITSP and one from an internal PBX. I've tried the following syntax but can't get it to work. Can someone clarify what I'm doing wrong? Here are the two gateway configs located in the Sip Profiles / External directory: <include> <gateway name="IPCM-VLAB-1"> <param name="proxy" value="10.10.3.10"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> <include> <gateway name="SDC1-VGW-1"> <param name="proxy" value="172.16.15.11"/> <param name="register" value="false"/> <param name="context" value="public"/> </gateway> </include> Here is the inbound route in the Dial Plan / Public directory (I'm just trying to get one number working at the moment): <extension name="From SDC1-VGW-1"> <condition field="destination_number" expression="^(2053148491)$"> <action application="bridge" data="sofia/external/IPCM-VLAB-1"/> </condition> </extension> I've also tried the gateway statement with no luck either: <action application="bridge" data="sofia/gateway/mygatewayname/555123456" /> Any ideas? Thanks, Ken Fulmer

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Cisco 7940 To Private
by Troy Anderson 30 Mar '10

30 Mar '10

I have some Cisco 7940's registered and working well with FS, but whenever I dial out from the Cisco phone, it indicates the call is "To Private" rather than showing the number I'm calling or something. I've put the SIP trace in pastebin http://pastebin.freeswitch.org/12589 - there is a Remote-Part-ID field that seems OK. Does anyone know what Cisco is looking for here? Thanks, Troy

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domain name in shared DB
by Madovsky 30 Mar '10

30 Mar '10

2 nodes in shared DB, node1 and node2 respectively with hostname node1.mydomain.org node2.mydomain.org the profile name is called mydomain.org which is shared by the 2 nodes. a user registers on node1 and wait calls. someone from external gateway call him, so user(a)node1.mydomain.org or user(a)node2.mydomain.org both it works well. but from user(a)mydomain.org it reaches the callee one time on three. my question is: can I avoid NAPTR SRV rules and replace it by a TCP load balancing server ? (if I use full TCP for SIP messages and UDP as normal after media negotiation) because it seems that NAPTR SRV work only if the caller proxy understands it. with a TCP load balancer I can avoid this, but not sure if it won't break the SIP protocol. Thanks Franck

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Snom wiki
by Klaus Hochlehnert 30 Mar '10

30 Mar '10

Just saw that the Snom interoperability page for FS was created/updated: http://wiki.snom.com/Interoperability/PBX/FreeSwitch Cheers, Klaus

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ringback with unsupported codec
by Fabio Pietrosanti (naif) 30 Mar '10

30 Mar '10

Hi all, i would like to provide ringback in early media using AMR that's supported only in passtrough. Would it possible, providing static AMR sample to be played back, to play it? Fabio Pietrosanti

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Bandwidth.com
by Sean Holt 30 Mar '10

30 Mar '10

Hello, Just received my bandwidth.com provisioning email with my sip_trunks and DID numbers. However, part of the instructions say I can¹t use a username and password when connecting to the gateway. I tried leaving the username and password blank in the gateway sip profile but it keeps giving me an error message, Error: username param is REQUIRED. Anyone ever setup a gateway with Bandwidth and how does one get around the username param requirement. Thanks Sean

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Invalid Boost Version 103 Expecting 100
by velusamy Krishnan 30 Mar '10

30 Mar '10

Dear all, Compiled the wanpipe with freeswitch and configured it. When I originate a call from the freeswitch command line it said the following error and got core dump. freeswitch@IVRFreeSWITCH> originate openzap/smg_prid/a/9940464753 &park() 2010-03-24 13:46:45.862315 [WARNING] sangoma_boost_client.c:246 TX EVENT (P): SYSTEM_RESTART:(87) [w1g1] Rc=0 CSid=0 Seq=0 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:355 Invalid Boost Version 103 Expecting 100 2010-03-24 13:46:45.862315 [CRIT] sangoma_boost_client.c:369 Critical Error: PQ Invalid Event lenght from boost rxlen=27 evsz=1027 NOTE: freeswitch version is 1.0.trunk ( 17084) wanpipe version is 3.5.8.6. smg_prid version is 1.63 Please help me......... Thanks..

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Re: [Freeswitch-users] Some question about mod_fifo ??
by Nguyễn Mạnh Hùng 30 Mar '10

30 Mar '10

Hi SeVen Du. Thank for your reply. About question 2: I think in real life (call center), supporters in queue must be known the real Caller_ID_number of customers. If caller_id_number is replated by 'fifo+<fifo_name>', caller_id_number will be invaluable informations. So, Is there anyway to preserve Caller_ID_number from the invaluable change as origination_caller_id_name=Queue,origination_caller_id_number='fifo+test_fifo'. Best regard ! > Hi all. > > I am newbegin and have 2 question as follow > > 1. Can config mod_fifo with member in external register ??? > > In my lab, i have OpenSIP as Proxy server and Register server and > Freeswitch as media server config mod_fifo (queue) > > User with extension 1010 is registerd with OpenSIPS server, queue 8000 is > media service in freeswitch server. Can i config mod_fifo in freeswitch > with member in Opensips server ? > > I try config in /fifo.conf.xml like > > > lag="5">{fifo_member_wait=nowait}user/1010@opensips_IP > > but it not work. > because it's not a valid dialstring. try sofia/ / instead of user/ it's only a dialstring, so, if you can make "originate " work then fifo should work. > Someone done this with opensips or another register server, please post some > guide. > > > > 2. With mod_fifo: caller_id_name and caller_id_number will be replated by > default config: > > origination_caller_id_name=Queue,origination_caller_id_number='fifo+test_fifo'. > So, if caller_id_number is 0123456.. it will be replates wit > 'fifo+test_fifo'. I want origination_caller_id_number has value is > 0123456.. > no, because you don't know which caller will actually been bridged to which agent before the agent actually answer the call. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Experience with FreeSWITCH & Turnserver.org ?
by RobertT 30 Mar '10

30 Mar '10

I chose this kind of setup, because I wanted to enforce ICE session negotiation between clients, that could possibly call from various kinds of NATs. sip_nat_detected tells you if the caller is residing behind NAT, but you don't know anything about another party yet. It might be hidden with NAT as well. Best regards, RobertT.

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encrypted RTP
by Woody Dickson 30 Mar '10

30 Mar '10

Hi, Is there anyway to transport RTP between FS and an endpoint in such a way that the the FS server is going to defined a key and only the endpoint that has the correct key can send and receive RTP from Freeswitch? Is there anyway to something that is similar? thanks, woody

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