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issue with quintum tenor and bridge_generate_comfort_noise
by Matthew Meek 27 Jun '13

27 Jun '13

We are migrating to using FS as an SBC between a telco and our Oracle Call Center Anywhere platform. Our current production calls are on a Quintum Tenor gateway using PRIs and we are moving to SIP via the SBC. The telco has a 5 minute RTP watchdog, and when Agents are working off-hook the RTP stream stops, requiring us to use bridge_generate_comfort_noise=true on the Agent leg. This works great as long as the inbound and outbound calls are both via SIP/Freeswitch. When the inbound call is via the Quintum Tenor (H323 to the CCA platform) and outbound is via the SIP/Freeswitch SBC, we get audio quality issues. The quality issue sounds like an overly-aggressive or poorly implemented VAD, where when the customer takes a small pause to talk, his or her voice is chopped for an instant when they begin speaking again. This is mostly apparent when the customer is reading off numbers... I do not understand why FS would be behave differently depending on the origin of the inbound call, as I am only adding bridge_generate_comfort_noise=true to the outbound call leg. Freeswitch is acting as a B2BUA SBC. I have packet captures on the outside interface of the SBC and can hear the audio issue in wireshark. Any thoughts? Thanks, Matthew Meek

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Odp: Re: can mod_httapi decide if the call should be answerred?
by Mariusz Czulada 27 Jun '13

27 Jun '13

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Dialing Group like local extension
by Kirk Wolf 27 Jun '13

27 Jun '13

I just installed freeswitch 1.2.10 for use in our small company. So far, most things are working very well. After skimming the "Freeswitch 1.2" book and the wiki, it was quite easy to get freeswitch built on a linode vm and connect to flowroute as a provider. I have an inbound DID that is being routed to a little IVR that I set up. In the IVR, a button routes to a "sales" group extension (2000). I'm using the sample dialplan/default.xml , which defines this extension as: <extension name="group_dial_sales"> <condition field="destination_number" expression="^2000$"> <action application="bridge" data="${group_call(sales@${domain_name})}"/> </condition> </extension> but I would like this to work with the same calling features as calling a normal user extension. For example, I don't get ring-back, and I want a ring-no-answer to go to a "group" voicemail box, etc. To do this, can I just copy all of the stuff from "Local_Extension"? This would add bind_meta_apps, ringback, and also a ton of other stuff that I don't understand. Of course, I would change the first bridge to make it a group call. IOW, can I do it like the following? Or are there applications here that should not be copied to a group-dial extension?  <extension name="*group_dial_sales*"> <condition field="destination_number" expression="*^2000$*"> <action application="export" data="dialed_extension=$1"/>  <action application="bind_meta_app" data="1 b s execute_extension::dx XML features"/> <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/> <action application="bind_meta_app" data="4 b s execute_extension::att_xfer XML features"/> <action application="set" data="ringback=${us-ring}"/> <action application="set" data="transfer_ringback=$${hold_music}"/> <action application="set" data="call_timeout=30"/>  <action application="set" data="hangup_after_bridge=true"/>  <action application="set" data="continue_on_fail=true"/> <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/> <action application="set" data="called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}"/> <action application="hash" data="insert/${domain_name}-last_dial_ext/global/${uuid}"/>  <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/> <action application="bridge" data="*group/support@${domain_name}*"/> <action application="answer"/> <action application="sleep" data="1000"/> <action application="bridge" data="loopback/app=voicemail:default ${domain_name} ${dialed_extension}"/> </condition> </extension> Any help very much appreciated. Kirk Wolf Dovetailed Technologies http://dovetail.com

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FreeSWITCH Dev's Salon and FreeSWITCH Training in China
by Seven Du 27 Jun '13

27 Jun '13

Hi there, Maybe I should announce this earlier but I was too busy. The Chinese Community is growing and we'll have a meeting at June 29 (GMT+8). http://www.freeswitch.org.cn/2013/05/16/freeswitch-cn-zhong-wen-she-qu-2013… And looks like we had sold out 100 free tickets at http://freeswitch-cn-2013.eventbrite.com/ . Contact me off list if you are happened to drop by and would like to have a coffee. We also offer a 1+2 days FS training follows the Salon: http://www.freeswitch.org.cn/2013/05/30/freeswitch-peixun-2013.html Thanks and would like to meet you all at ClueCon this year. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com

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Lost and delayed DTMFs
by Jack 27 Jun '13

27 Jun '13

We are having troubles with lost and delayed DTMFs with PhonePower, one of our trunk providers. I was wondering if there is a setting in freeswitch that will force the provider to send inband only. Or if that is even a viable solution?? thanks, Jack

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can mod_httapi decide if the call should be answerred?
by Rafal Gwizdala 27 Jun '13

27 Jun '13

Hi, I've got a simple question to you: can I use httapi before a call is answerred and, for example, hang up/reject the call if my application decides it doesn't want to handle it? I'd like to give the caller a 'busy' signal as a feedback or something like this. Thanks, RG

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mod_spandsp: no modems available
by Peter Steinbach 27 Jun '13

27 Jun '13

Hello, according to the Hylafax wiki page we could set up 4 modems with Hylafax. We also used termnetd + interceptty to extend the modem interfaces to a remote Hylafax server. The interface works outbound: Fax from Hylafax to Freeswitch works. However bridging an incoing fax to modem/a/a fails very often. I we want to send a fax, right after the processes are started and modems are connected, it always fails with "no modems available". The same happens if we explicitely set an interface number, e.g. modem/1/a, which is available at that time. However, after waiting a while, we can get a call through, an available modem is detected. It seems as if modems are detected, but they are not considered as available. Just after some time they may be released, in my opinion. My questions: * has anybody else had this issue? * does anybody know, how to workaround this? * maybe this is some behaviour of termnetd + interceptty? * what's the best way to debug this? I have not seen any Freeswitch command for mod_spandsp. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Caller-Destination-Number truncated
by Nageshwara Rao Moova 27 Jun '13

27 Jun '13

Hi All, Recently i updated my freeswitch and see that "Caller-Destination-Number" only passes last two digits of DNIS. I have B2B setup, earlier when driver passed DNIS as 2001 fresswitch mod_freetdm variable "Caller-Destination-Number" had 2001. But now with latest version of freeswitch, the Caller-Destination-Number has 01, what is causing this truncation when Caller-Source is mod_freetdm. But with Caller-Source mod_sofia (VoIP) we don't have any issue with Caller-Destination-Number. Any help appreciated. -- regards & thanks -- m nageshwara rao 99891 86280

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Two FreeSWITCH boxes talking to each other
by Stanislav Sinyagin 27 Jun '13

27 Jun '13

I'm not sure if I need to open a bug for this: Two FreeSWITCH boxes, 1.2 stable, * A is the PSTN gateway and has a user x(a)domain.com (domain has an SRV record pointing to the box port 5080) * B has a gateway in its sip_profiles pointing to the user at A. First, I configured B this way:  <gateway name="x(a)domain.com"> <param name="username" value="x"/> <param name="realm" value="domain.com" /> <param name="password" value="XXXXX"/> <param name="extension" value="731"/> <param name="expire-seconds" value="3600"/> <param name="register" value="true"/> <param name="register-transport" value="udp"/> <param name="retry-seconds" value="30"/> <param name="caller-id-in-from" value="true"/> <param name="ping" value="27"/> </gateway> Then when from A I wanted to call the user x, A failed, and sofia debug showed that it tried to resolve a record that is a combination of B's IP address and the domain name. A did not even try to send anything toward B. After changing to <gateway name="x_domain.com"> it started working as expected. shall I open a bug? I cannot provide logs because they contain confidential information. regards, stan

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[newbie] route incomming call to endpoint using fs_cli
by Raymond Domingo 27 Jun '13

27 Jun '13

Hello, I'm just taking first steps in learning freeswitch. What I would like to do is route my incomming call to 'some' endpoint using fs_cli. (just to see how all is working). My config (snapshot): <extension name="raymondtest"> <condition field="destination_number" expression="^5800$"> <action application="answer"/> <action application="sleep" data="200000"/> </condition> </extension> <extension name="echo"> <condition field="destination_number" expression="^9196$"> <action application="answer"/> <action application="echo"/> </condition> </extension> Both endpoints are working. So when I call using my softphone to 5800 I can see the call in fs_cli using show calls: 0dbf4066-dd7a-11e2-a52c-b9fe005eb6ff,inbound,2013-06-25 11:31:46,1372152706,sofia/internal/1500@172.30.4.186,CS_EXECUTE,1500,1500,172.30.4.132,5800,1500@172.30.4.186,,ACTIVE,,,,,fs-devel,,,,,,,,,,,,,,,,,,,,, But how can I route this call USING FS_CLI commands to the echo endpoint ? I looked into originate, bridge and sofia. But I couldn't find out how to do this. Best regards, Raymond Domingo -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/newbie-route-incomming-call-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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