May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please)
by Peter Olsson 21 May '12

21 May '12

I guess it mostly depends on the Polycom device. For instance, it's a big difference if the call is transferred using blind or supervised transfer. You will probably need to collect the SIP traffic, and check what actually happens when doing this scenario. Mvh Peter Olsson Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För ghallab Skickat: den 21 maj 2012 12:07 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please) Hi guys, Can you explain me what well happen if I set the variable "ignore_display_updates=true" before I make bridge action and what will happen if I set it to false? because I have a problem with FS and Polycom 550: when A call from outside of the FS network to B which is inside, then B answer and transfer the call to C which is inside also, when C's phone is ringing he see B's caller ID and when he answer, he see A's caller ID. Me I want that he see always caller ID of A. I tried to set "ignore_display_updates=true" in this case B see just C's caller ID when his is ringing and when he answer. There is any way to fix this issue? I am thinking about to set "effective_caller_id_number={$origination_caller_id_number}" but I am not sure if this will work. !DSPAM:4fba049932768390320688!

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Re: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please)
by Peter Olsson 21 May '12

21 May '12

I'm not sure how Polycom does their transfers, but usually these are the differences; When doing blind transfer, the phone has one call leg, and then sends a SIP REFER (to FreeSWITCH in this case), telling it to take that call and blindly transfer it to another extension. In this case the original CallerID should be transferred correctly. When doing supervised/consultation transfer, the phone will start by making a new call to the extension you want to transfer to. In this state the call will be from the phone doing the transfer, to the phone that is the destination (and the caller-id on the destination phone will be the transferers phone). When the destination phone picks up, the transferer says something like "Ok, here is a call for you", and then completes the transfer. Now it will probably send a REFER with replaces tags to FreeSWITCH, telling it to drop the two call legs to the transferer, and instead directly bridge the incoming external call to the transfer destination (these two legs already exist, so FS will just start bridging the RTP for these channels). As soon as this has happened, FS will send a display update, since the CallerID actually changed now, from the transferers phone to the original external call. As I said, I'm not sure if this is how Polycom does their transfers, but the basic difference between a blind and supervised transfer is as described above. So in this case, if you want to do a supervised transfer, and let the transfer target get the original callerID directly (before the call are actually transferred), I think you will need to check out the entire SIP call flow, and then take appropiate actions (either change config in FS, and/or change config on the phones). However, most people expect it to work as it does by default... /Peter Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För ghallab Skickat: den 21 maj 2012 14:18 Till: FreeSWITCH Users Help Ämne: Re: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please) Could you tell me what is the impact of each one ( transferred using blind or supervised transfer)? On 05/21/2012 09:22 AM, Peter Olsson wrote: I guess it mostly depends on the Polycom device. For instance, it's a big difference if the call is transferred using blind or supervised transfer. You will probably need to collect the SIP traffic, and check what actually happens when doing this scenario. Mvh Peter Olsson Från: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] För ghallab Skickat: den 21 maj 2012 12:07 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] ignore_display_updates meaning an polycom problem (urgent please) Hi guys, Can you explain me what well happen if I set the variable "ignore_display_updates=true" before I make bridge action and what will happen if I set it to false? because I have a problem with FS and Polycom 550: when A call from outside of the FS network to B which is inside, then B answer and transfer the call to C which is inside also, when C's phone is ringing he see B's caller ID and when he answer, he see A's caller ID. Me I want that he see always caller ID of A. I tried to set "ignore_display_updates=true" in this case B see just C's caller ID when his is ringing and when he answer. There is any way to fix this issue? I am thinking about to set "effective_caller_id_number={$origination_caller_id_number}" but I am not sure if this will work. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fba232f32763870889139!

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Advance SBC
by Miha 21 May '12

21 May '12

Hi, I am looking on wiki about implementing SBC with mod_easyroute and mod_lcr (http://wiki.freeswitch.org/wiki/Advance_SBC_with_mod_lcr_and_mod_easyroute) I have configured as is written on wiki in everything works. Just one question. Should mod_lcr automatically be routing to less cost destination if first destination is down (this does not happend)? Regards, Miha

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ignore_display_updates meaning an polycom problem (urgent please)
by ghallab 21 May '12

21 May '12

Hi guys, Can you explain me what well happen if I set the variable "ignore_display_updates=true" before I make bridge action and what will happen if I set it to false? because I have a problem with FS and Polycom 550: when A call from outside of the FS network to B which is inside, then B answer and transfer the call to C which is inside also, when C's phone is ringing he see B's caller ID and when he answer, he see A's caller ID. Me I want that he see always caller ID of A. I tried to set "ignore_display_updates=true" in this case B see just C's caller ID when his is ringing and when he answer. There is any way to fix this issue? I am thinking about to set /"effective_caller_id_number={$origination_caller_id_number}" but I am not sure if this will work. /

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LINGER
by Bernard Fluixa 19 May '12

19 May '12

Hello, I am working on an outbound ESL application in C on Linux. I can send commands and receive events normally. My problem is that the "linger" command returns the "-ERR not controlling a session" error message. The socket application is set to be in async full mode. (<action application="socket" data="127.0.0.1:9090 async full"/>). Any idea? Thank you B

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Huawei E169 Group Buy
by Brian Foster 19 May '12

19 May '12

Fellow freeswitchers, I am sending this email out to let you all know that I am setting up a group buy for the Huawei E169. It should be compatable with mod_gsmopen, and that will be tested before we go and do a buy. The reason why there is interest in this particular model is the fact that it does have an external antenna jack. Otherwise they are fairly close to the E1550. If you are interested, make sure you do your research to make sure this single will work for you. No refunds. Send an email to gsmdongle(a)endigotech.com letting me know how many you want and the best way to get ahold of you. Deadline is May 30th to get your requests in. There will be no refunds. There will be no markup on price. Buyers are responsible for shipping. Price range should be $25-35. US/Canada residents only. Must be 18 years or older. Payment via PayPal only. Notice will be sent out a few days after May 30th to confirm we have enough interested and how much each single will be. At that time you will need to send a payment to the PayPal address. Payments will be in two stages, the first will be for the product itself, the second will be for the shipping to your address. There needs to be a minimum of 20 dongles ordered before I can go ahead and buy. Payments need to be collected prior to me going ahead and buying. If you have any questions, email me off list. -BDF

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bridge_early_media and bypass_media
by Adrian Andrei 19 May '12

19 May '12

Hello, In my dial plan I use bypass_media set to true because I want the codec negotiation be made by end points. Also I want the early_media to be bridge from Bleg to Aleg. For this setup I design this dialplan: <condition field="destination_number" expression="^1"> <action application="set" data="bypass_media=true"/> <action application="set" data="bridge_early_media=true"/> <action application="bridge" data="sofia/external/xxxx"/> </condition> Setting bypass_media and bridge_early_media to true cause no early media bridge. If I set only bridge_early_media=true and uncomment bypass_media line then early media is bridged but no codec negotiation between end points. How can I have also codec neg. between end points and early media bridged? Thank you

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local_ip_v4 is resoved with 127.0.0.1
by Sharad Garg 19 May '12

19 May '12

Hi All Just installed the freeswitch with centos. I found that freeswitch is getting the local_ip_v4 = 127.0.0.1 & thats why the calls are not handled by freeswitch. I have set auto detection. With the same centos / freeswitch, I checked it up with another pc & found everything working. This time local_ip_v4 is getting the ip of eth0. In both scenerio, pc is having eth0 & eth1. >From this excercise, I concluded that there is something wrong in installation at OS side. But I do not know what is wrong. Actually I do not want to configure the eth0 IP manually in xml files. I want auto detection only. So could someone help me to diagnose this issue ? Thanks in advance. regards Sharad

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Fwd: RTP stats explaination
by Arif Hossain 19 May '12

19 May '12

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all, We are getting very poor quality of voice during testing of a new filtering application of us. The application receives packets from kernel using netfilter_queue library. Then insert the packets into a new user managed queue and does some transformations on it, like concatenation of udp payload. The network is healthy. Its inside our lab. And it does not drop packets or anything . In our app we do not forward packet immediately. After enough packet received to increase rtp packetization time (ptime) the we forward the message over raw socket and set dscp to be 10 so that this time packets can escape iptable rules. >From client side the RTP stream analysis shows nearly every stream as problematic. summery for some streams are given below : Stream 1: Max delta = 1758.72 ms at packet no. 40506 Max jitter = 231.07 ms. Mean jitter = 9.27 ms. Max skew = -2066.18 ms. Total RTP packets = 468 (expected 468) Lost RTP packets = 0 (0.00%) Sequence errors = 0 Duration 23.45 s (-22628 ms clock drift, corresponding to 281 Hz (-96.49%) Stream 2: Max delta = 1750.96 ms at packet no. 45453 Max jitter = 230.90 ms. Mean jitter = 7.50 ms. Max skew = -2076.96 ms. Total RTP packets = 468 (expected 468) Lost RTP packets = 0 (0.00%) Sequence errors = 0 Duration 23.46 s (-22715 ms clock drift, corresponding to 253 Hz (-96.84%) Stream 3: Max delta = 71.47 ms at packet no. 25009 Max jitter = 6.05 ms. Mean jitter = 2.33 ms. Max skew = -29.09 ms. Total RTP packets = 258 (expected 258) Lost RTP packets = 0 (0.00%) Sequence errors = 0 Duration 10.28 s (-10181 ms clock drift, corresponding to 76 Hz (-99.05%) Any idea where should we look for the problem? - -- - -aft -----BEGIN PGP SIGNATURE----- Version: OpenPGP.js v0.1 Comment: http://openpgpjs.org wsBcBAEBAgAQBQJPtiJ7CRCJVJ6A/SK8awAAHJgH/jRURuZWygqJTX7zafjK 807MvWxODhtw2p4ZLg+sCNJ0OpZlV5GftPKwH9XB4GfUKhmTPfCZ3n8vF6PU nWKMdBDM3D/K+DktpiGuAK90A5lwniFptN8DGt/ltqdAaSAlGbz1E6LGGdlw 1w/6lPeO3J9HKpkNQA2nfn3q/s5ZHQiYoGvgFofkm85LBSYgyn4b7YDY7qnp wtsGkSR1h66+q37Hb+1iZyWrHj26YJ+sNRBe8GzKdEccwrKMX1gJFNl5BBfl frGsZ3Y02cSYFg5L1vVstl8F/yTz8JXTGk6yKbtA0N+TRhI7gIJ/VItm1SJt JcF94ylPjIWiw9Iacv8ec88= =jiIm -----END PGP SIGNATURE----- -- -aft

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Calls to the external phone
by Dubrovskiy Viacheslav 19 May '12

19 May '12

Hi. Using the default configuration. There are internal and external real IP. I have several registered account from the internal network. I want to connect remote user from Ethernet. For this added external real IP to the external sip profile (param name="sip-ip" and param name="rtp-ip") and copied the directory/default.xml in directory/externa.xml which changed the <domain name="external_IP"> Now remote users can register and make calls. But there's a problem: No one can call to this external user. Tell me, what am I doing wrong? Thank you -- WBR, Dubrovskiy Viacheslav

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