April 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to configure media bypass when using mod_callcenter
by fieldpeak 10 Apr '12

10 Apr '12

Hi Maestros, i'm using mod_callcenter to realize a simple senario, pstn user call to media gw, then go to queue number 7000 of fs, then fs hunt a idel agents(registered users) to accpet the call. it works fine. however, i would like to make the media bypass the fs after the agent answer the call, i.e. media-gw<->agent, however, from the trace,the media still go via fs. from the trace, i saw when fs make the call to agent, the fs initiate a new "INVITE" with itself IP address as the connect ip address in SDP, which caused the bypass failed. (in the attahed trace, .159 is the fs, .50 and .117 are two registered users). i have configure below, and if two agent call for each other directly, the media did bypass fs. internal.xml <param name="inbound-bypass-media" value="true"/> <param name="inbound-proxy-media" value="false"/> othe configuraton as following, dial plan: <extension name="7000_Queue" > <condition field="destination_number" expression="^7000$"> <action application="answer"/> <action application="playback" data="/usr/local/freeswitch/sounds/custom/8000/transfer_to_agent.wav"/> <action application="callcenter" data="7000_Queue@localhost"/> <action application="system" data="mkdir -p $${base_dir}/recordings/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/"/> </condition> </extension> callcenter.conf.xml: <configuration name="callcenter.conf" description="CallCenter"> <settings>  </settings> <queues> <queue name="7000_Queue@localhost"> <param name="strategy" value="round-robin"/> <param name="moh-sound" value="$${hold_music}"/> <param name="time-base-score" value="queue"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="0"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> </queue> </queues>    <agents> <agent name="agent1001@localhost" type="callback" contact="[call_timeout=10]user/1001@$${domain_name}" status="Available" /> <agent name="agent1002@localhost" type="callback" contact="[call_timeout=10]user/1002@$${domain_name}" status="Available" /> <agent name="agent1003@localhost" type="callback" contact="[call_timeout=10]user/1003@$${domain_name}" status="Available" /> </agents>  <tiers> <tier agent="agent1001@localhost" queue="7000_Queue@localhost" level="1" position="1"/> <tier agent="agent1002@localhost" queue="7000_Queue@localhost" level="1" position="1"/> <tier agent="agent1003@localhost" queue="7000_Queue@localhost" level="1" position="1"/> </tiers> </configuration> Could you please any one help, thanks a lot! BR, Charles

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Acoustic Echo Cancellation on Freeswitch side
by Imran Moinuddin 10 Apr '12

10 Apr '12

Hi there, I was wondering whether there were any echo cancellation alternatives available that could potentially work on the Freeswitch server side. This is mainly to cater for people that dial into my service using just their laptop microphone + speakers and their client app doesn't support echo cancellation. I realize that this could be an expensive proposition (from a computational perspective) but curious if the option (commercial or otherwise) exists. Thanks! Imran

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Developers is this a bridge change or Jira/Bug issue?
by Mario G 10 Apr '12

10 Apr '12

Testing the recent gits introduced a problem which I was able to work around but wondered if this is an intended change or bug: If the caller_id_number is one of the numbers in a bridge command only 1 ring is done and the bridge ends, it appears as if originate_timeout is ignored, the trace is normal. Prior to the git update the bridge would continue to ring to the timeout and the caller would be bridged to all the numbers in the bridge so any phone could answer the call. What used to work in older git: <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222:_:sofia/gateway/${dial_gateway}/19163334444"/> Newer git. Now I have to execute an extension to check who is calling so they are not in the bridge: <extension name="bridge_call"> <condition field="destination_number" expression="^bridge_call$"> </condition> <condition field="${caller_id_number_work}" expression="^9161112222$" break="on-true"> <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19163334444"/> </condition> <condition field="${caller_id_number_work}" expression="^9163334444$" break="on-true"> <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222"/> </condition> <condition field="${caller_id_number_work}" expression="^(\d+)$"> <action application="bridge" data="<originate_timeout=23,origination_caller_id_number=${caller_id_number},origination_caller_id_name=${caller_id_name}>group/home@${domain_name}:_:sofia/gateway/${dial_gateway}/19161112222:_:sofia/gateway/${dial_gateway}/19163334444"/> </condition> </extension>

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dinamically change sip_from_user in the dialplan
by Nestor A Diaz 09 Apr '12

09 Apr '12

Hi Guys, I want to know if is possible to change the sip_from_user at the dialplan. My current pstn sip provider needs the caller id to be set to one of the numbers assigned to me, this way i can place the call using the following options at the dialplan: <extension name="local"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=12345678"/> <action application="bridge" data="sofia/profile/$1(a)sipprovider.com"/> </condition> </extension> The above works, also if i define a gateway as the following: <include> <gateway name="myprovider"> <param name="from-user" value="12345678"/> <param name="proxy" value="sipprovider.com"/> <param name="register" value="false"/> <param name="ping" value="25"/> </gateway> </include> and place the call using: <action application="bridge" data="sofia/gateway/myprovider/$1"/> the above woks too, but i am tied to the from_user i have defined. So i wanted to use be able to change the from_user dinamically according to my needs at the dialplan level, is that possible ? i have tried the following but it doesn't work <action application="bridge" data="{sip_from_user=12345678}sofia/gateway/myprovider/$1"/> Any suggestions ? or is not possible to define a gateway and change some of its parameters in the dialplan ? i just want to use the ping functionality and have something more organized. Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211@tiendalinux.com Email/MSN: nestor(a)tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia

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Help with tone detection
by Jean-Marc Hyppolite 09 Apr '12

09 Apr '12

Hello everyone, Is it possible to have some help in setting up mod_spandsp for call progress tone detection? I have not been successful thus far. I keep getting an "error reading frame" message like this one below. ============================================================================ =================== 2012-04-09 16:03:22.550303 [DEBUG] mod_spandsp_dsp.c:379 (sofia/internal/sip:1000@184.162.247.104:32986) Starting tone detection for '1' 2012-04-09 16:03:22.550303 [INFO] mod_spandsp_dsp.c:411 (sofia/internal/sip:1000@184.162.247.104:32986) initializing tone detector 2012-04-09 16:03:22.550303 [DEBUG] switch_core_media_bug.c:462 Attaching BUG to sofia/internal/sip:1000@184.162.247.104:32986 2012-04-09 16:03:22.590301 [DEBUG] switch_core_io.c:340 Setting BUG Codec PCMU:0 2012-04-09 16:03:22.590301 [INFO] mod_spandsp_dsp.c:422 (sofia/internal/sip:1000@184.162.247.104:32986) error reading frame 2012-04-09 16:03:22.590301 [INFO] mod_spandsp_dsp.c:447 (sofia/internal/sip:1000@184.162.247.104:32986) destroying tone detector ============================================================================ =================== My spandsp.conf.xml file: ============================================================================ =================== <configuration name="spandsp.conf" description="Tone detector descriptors"> <descriptors>    <descriptor name="1"> <tone name="CED_TONE"> <element freq1="2100" freq2="0" min="500" max="0"/> </tone> <tone name="SIT"> <element freq1="950" freq2="0" min="256" max="400"/> <element freq1="1400" freq2="0" min="256" max="400"/> <element freq1="1800" freq2="0" min="256" max="400"/> </tone> <tone name="REORDER_TONE"> <element freq1="480" freq2="620" min="224" max="272"/> <element freq1="0" freq2="0" min="224" max="272"/> </tone> <tone name="BUSY_TONE"> <element freq1="480" freq2="620" min="464" max="516"/> <element freq1="0" freq2="0" min="464" max="516"/> </tone> </descriptor> </descriptors> </configuration> ============================================================================ =================== Thank you for your help. Jean-Marc.

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SCA/SLA not working with Polycoms on inbound calls
by Vik Killa 09 Apr '12

09 Apr '12

I have SCA/SLA working with Polycom phones (firmware 3.3.1) on outbound calls. But inbound is not working. I have read over http://wiki.freeswitch.org/wiki/Shared_Line_Appearance and followed the instructions but still no luck. I have the following in my sofia profile: <param name="manage-shared-appearance" value="true"/> <param name="multiple-registrations" value="true"/> <param name="manage-presence" value="true"/> I have 2 Polycom phones registered (1000 & 1004) and the second line on each phone is a shared line, 9000. A call comes in to 9000 and both phones ring as expected but once one of them picks up, the line 2 on the other phone has no affect on the call (normally it would allow 'barge-in' or it would take the call if it was on hold). On outbound, everything works so I do not believe it is a Polycom configuration issue. Here is the dialplan bridge I'm using: <action application="bridge" data="${sofia_contact(9000(a)thedomain.com)}"/> Any suggestions would be greatly appreciated, thanks.

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BLF Presence
by DJB International 09 Apr '12

09 Apr '12

I currently have my Polycom 650 phone set up with presence enabled: <feature feature.1.name="presence" feature.1.enabled="1" /> However, when I looked at the table sip_presence in sofia_reg_internal.db, I do not see any information in there. What I am trying to achieve is something similar to "sip show subscription" to verify BLF subscriptions. Is there something that I am missing here. Please advise. Thank you.

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alternative to mod_hash which is persistent
by Fahrwerk Kurierkollektiv (Technik) 09 Apr '12

09 Apr '12

Hi all, As the wiki writes, that mod_hash "provides a non-persistent datastore for limit data. This data is specific to a particular FreeSWITCH instance", I'm searching for an alternative. I would like to setup a callforwarding extension, and using mod_hash at the moment for storing the target number. After a reboot, of course, the hash is gone. Could someone point me at an alternative? Thanks, Georg

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mod_avmd false positives, ineffecient
by David Brazier 09 Apr '12

09 Apr '12

Hi I've been testing mod_avmd and got a lot of false positives - i.e. normal speech detected as beep. As I understand it, the basic approach is to look for audio with one very dominant frequency - the beep. It does this by getting a frequency estimate and then calculats the amplitude of the audio at that frequency, looking for a consistent run of high amplitudes. But I've found a few things that are either not right or I have misunderstood: 1. The expensive desa2 frequency estimate is calculated on every sample, and worse, as each frame arrives (say 160 samples) it calculates the desa2 for every sample in the circular buffer - 800 samples (rounded up to 1024 I think?). So that is using excessive CPU, and anyway I don't think the frequency estimate should be calculated on every sample, because in normal speech, the frequency is changing and so successive samples might register a high amplitude but with different frequencies, causing false positives. 2. The amplitude function doesn't appear to produce a constant result when the input audio is a pure tone, although the deas2 frequency estimate is close. 3. The running mean sma_b is tested against zero - which basically means any positive amplitude meeting the "valid" cases will count as a detected beep. I have re-worked mod_avmd.c to: 1. Calculate the frequency only once every 10ms (sine len). 2. Calculate the running variance of frequency over the required beep length (100ms) - ie. last 10 estimates. This is producing good results - would welcome comment & if anyone else can try it that would be great. Could be further tided up as it doesn't really require a circular buffer for samples at all - could just do the desa2 calculation direct from the sample buffer, at least once per frame (2 per frame for 8000kHz, 20ms frames). Also, goertzel.c isn't used by the module at all, and amplitude.c isn't used in my version and can be removed. Raised as http://jira.freeswitch.org/browse/FS-4090 with proposed patch. Regards David

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(no subject)
by Manjiri Deshpande 09 Apr '12

09 Apr '12

http://elastomer-kazan.ru/entrance.php?wuluhaxynj=56&jvabyxuv=43

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