November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

rtp-timer-name issue and question
by Matt Stockton 19 Nov '11

19 Nov '11

Hi all, I am battling the following issue: My applications are primarily lua scripts that play files, accept DTMF , and make db calls and url callbacks to post information. At one point, I was having issues with an ever increasing delay for DTMF recognition as the application progressed (e.g. I would use playAndGetDigits, and the DTMF wouldn't be 'recognized' for a delayed amount of time. This time would increase and it appeared to be dependent on the number of db calls and url callbacks I made in the app -- as I made more, the delay would get worse). I asked around, and ended up changing this setting in my external sip profile: rtp-timer-name = none (it was soft before). This fixed my issue. Now, I am battling a separate issue, and I seem to have isolated it's cause to setting rtp-timer-name to none instead of soft. The problem now is that when I want to play an mp3 file, and I use streamFile with shout or playback with http_get, the beginning of the file is not played at all (playing starts probably about 200-400ms into the file. This is very repeatable for me - it's not happening to every streamed file, but it seems to be very deterministic on while line of lua script it's happening (the streamFile is happening in a loop, and it seems to always be happening to the first streamed file). If I change rtp-timer-name to soft, everything is fine, but then I'll have the first problem I described above. So, to fix one problem I need rtp-timer-name = none, for the other problem, I need rtp-timer-name = soft. I must be doing something wrong here, any suggestions? Thanks, Matt

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Custom Call reporting
by Frank Busalacchi 19 Nov '11

19 Nov '11

Hello, and thanks for a great piece of software! I currently have FreeSwitch recording all inbound calls. I'm envisioning writing an intranet webpage that would allow managers to be able to generate a list of hyperlinks to call recordings based on ad hoc criteria such as an extension that was on the call path , CLID of the inbound call, date etc. I'm figuring the best way to do this is to name the inbound call recording by its UUID, and by parsing the logs and extracting the UUID of all calls that match the criteria. The system is currently configured to log through CDR-CSV with default installation settings. When I peruse the logs, I see only the beginning of the phone call (Initial inbound, or initial outbound)....What I don't see is the entire "call path". How do I get the log to show every transfer of the call etc? My thought is that if I could do that, then I could basically grab the UUID of the call, grab all log entries that have the UUID in it, and see every extension that was involved with that UUID. Am I on the right path? I'm tinkering with a production system, so I don't want to mess with settings and interrupt the business during business hours. My guess is that I need to set the CDR-CSV log setting to log both AB instead of the default to get the info I am looking for to show up in the log? Thanks, -Frank

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Transcoding and TLS Handling Proxy
by Adam Sherman 18 Nov '11

18 Nov '11

Good Morning, I have an existing SIP-based hosted PBX service running on a commercial platform. To handle some edge cases, I would like to setup a proxy to achieve two purposes: transcoding from iLBC, SILK & G.722 to G.711 and offering SIP over TLS and SRTP to customers that require it. I was originally looking to see if OpenSIPS could do this, but I now believe this is better achieved by a B2BUA and thus FreeSWITCH (please comment on that logic). Some key points I have found, so far: 1. Since the proxy will be transcoding and/or handling SRTP streams, it will also need to handle NAT traversal; 2. It should be completely stateless, I think; 3. UACs will be registering through it (they will be configured with it as an outbound proxy); 4. If one UAC ends up being connected to another UAC (e.g. extension to extension dialling), we want the SRTP stream to simply be relayed; 5. This proxy will simply relay all signalling and decrypted/transcoded media to the existing switch; 6. If I run into scaling issues, I can use Sangoma's hardware transcoding solution with FreeSWITCH; It would be great to hear all your comments and suggestions of what components and modules to use and any other tidbits you can provide. Also, if anyone has done something like this already and wants to offer commercial services to replicate it for me, send me an email. Thanks, A. -- Adam Sherman Technologist Deputy SARCOM, SAR Global 1 Coordonnateur de L'AQBRS, région 07 Outaouais +1 613 797 6819

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Create User xml file automatically (directory/default/called_no.xml)
by Sanath Prasanna 18 Nov '11

18 Nov '11

Hi all, I need to do following in voicemail module of FS. when called number is not available, call is routed to voicemail. I did that using dial plan.But system error coming as "user not exist". When I create directory/default/called_no.xml file manually, it is not coming. I need to create that file automatically for each called number, which is not available. How to do that ? Br, Sanath

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external sip profile
by Chad Vogel 18 Nov '11

18 Nov '11

hello, I'm trying to switch from asterisk to freeswitch; however i'm wondering how can I create a sip profile because the sip profile i created doesn't seem to function with level 3. here is the sip profile i created that isn't working: <include> <gateway name="Level3"> <param name="username" value="USERNAME"/> <param name="password" value="PASSWORD"/> <param name="proxy" value="4.55.35.60:5070"/> <param name="expire-seconds" value="3600"/> <param name="register-transport" value="udp"/> <param name="register" value="true"/> </gateway> </include> here is my asterisk profile (it works): [level3_out] type=peer nat=no host=4.55.35.60 username=***Username*** secret=***Password*** dtmfmode=rfc2833 port=5070 [level3_in] nat=no insecure=very dtmfmode=rfc2833 disallow=all context=from-trunk canreinvite=no allow=ulaw&alaw host=4.55.35.60 type=peer port=5070 How can I create a sip profile that will function the same in freeswitch?

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How to tell Leg A to still go on with it's executable action lists after an transfer/REFER?
by Raimund Sacherer 18 Nov '11

18 Nov '11

Hello to all, Is it true that a call looses the rest of the dialplan actions it has currently after I do a transfer/REFER? It seems so. How can I force the execution of more dialplan actions after the transfer/REFER? What it does now is: What works: A -> call in -> B Accepts -> B Talks -> B Hangs up -> A is transferred to IVR What does not work: A -> call in -> B Accepts -> B Talks -> B Transfers/REFER to C -> C Talks -> C Hangs Up -> A should be transferred to IVR It seems that A looses everything it should do after B (or C) hangs up, which had been defined in the dialplan, after it got transfer/REFERED to C. How do I rectify this? exports? variables? ?? Thank you, best regards Ray --

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Anyone using t38modem with FS (gatewaying from audio<>t38) and Hylafax successfully?
by Alex Crow 18 Nov '11

18 Nov '11

Hi list, I am trying to figure out, after abandoning trying to get iaxmodem working in FS, if it is possible to use t38modem to connect FS to Hylafax, mainly for the purpose of sending and receiving non-t38 faxes via an ISDN card or non-t38-aware gateway (I'm currently using our Mitel 3300 as the gateway and voice calls between the two systems are fine). I have managed after a lot of effort to get t38modem compiled and running, and it seems to react to incoming fax calls coming in via the Mitel gateway from a normal fax machine, but the fax machine does not hear any reply to its initiation tones and the call hangs up in the end. Ivan on this list has suggested that I don't really need to do the gatewaying as t38modem can work with audio over SIP as well, so I tried removing all the t38 lines from the dialplan but with no success. My fax dialplan looks like this: <extension name="from_mitel" > <condition field="network_addr" expression="10.10.0.2"/> <condition field="context" expression="public"/> <condition field="destination_number" expression="(^1001$)"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="execute_on_answer=t38_gateway peer"/> <action application="bridge" data="user/t38modem@${domain_name}"/> </condition> </extension> t38modem is invoked thusly: t38modem -tttt -o /var/log/t38modem.log --no-h323 -u t38modem --sip-register t38modem(a)192.168.20.245,password --sip-listen udp$127.0.0.1:6060 --ptty +/dev/ttyT38-1 --route "modem:.*=sip:<dn>@192.168.20.245" --route "sip:.*=modem:<dn>" Where 192.168.20.245 is the ip of the FS box. Any ideas? I can provide logs if required. Thanks, Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)

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a=fmtp missing on FS answer
by Camila Troncoso 18 Nov '11

18 Nov '11

Hello, I´m having some problems with códec negotiation and one way audio . I take some Sip captures and I notice that Freeswitch is not inserting * fmtp* parameters in SDP in his 200OK response. I notice that this is a common behavior. I really need that FS generates the complete SDP (with fmtp) because some devices only accept for example G729 annexb=no. I tried with the function sip_append_audio_sdp but it only accepts one a=fmtp line. “ <tr valign=top> <th align="left"> @RFC3264: SDP Offer/Answer Negotiation </th> <td> Generating and processing offers or answers. </td> <td> - "a=fmtp" parameters are not taken into account when generating or processing answer </td> </tr> ” Regards, *Camila Troncoso **|* Ingeniero de Desarrollo RedVoiss *|*ctroncoso(a)redvoiss.net Santiago - Chile *|* +56 2 2408535 www.redvoiss.net

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A in B leg CDR still sending after value=fals
by Miha Zoubek 18 Nov '11

18 Nov '11

HI, I have problem with modul mod_radius_cdr and dial plan. I have try to disable a and than b leg in my dial plan, but freeswitch is still sending values. Please tell me what I am doing wrong. Dial plan: <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/> <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/>  <action inline="true" application="set" data="CALLINGNUMBER=${caller_id_number}"/> <action inline="true" application="set" data="CALLINGNUMBER=${destination_number}"/> <action inline="true" application="set" data="USERNAME=${dialed_extension}"/> <action inline="true" application="set" data="PASSWD=$${default_password}"/>  <action application="set" data="process_cdr=false"/> <action application="export" data="_nolocal_process_cdr=false"/> <action application="sleep" data="2000"/> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> <action application="export" data="dialed_extension=$1"/> <action application="bridge" data="user/0${dialed_extension}@${domain_name}"/> <action application="set" data="process_cdr=false"/> <action application="export" data="_nolocal_process_cdr=false"/> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/> <action application="log" data="INFO billing_model=${billing_model}"/> <action application="log" data="INFO credit_amount=${credit_amount}"/> <action application="log" data="INFO currency=${currency}"/> <action application="log" data="INFO preffered_lang=${preffered_lang}"/> <action application="log" data="INFO credit_time=${credit_time}"/>  <action application="log" data="INFO return_code=${return_code}"/>  </condition> </extension> Regards, Miha

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Call transferring from analog phone is not working (OpenZap)
by George D'mithrov 18 Nov '11

18 Nov '11

Hi all, Am new to freeswitch. I configured freeswitch with OpenZap (analog lines). Now am able to make call between SIP phone and Analog phone. But call transferring is not working from analog phone (am using hook flash, blind transferring), but the same is working from SIP phone. when a call is transferring from the analog line the actual call is get disconnected. Any suggestion is well appreciated, Thanks in advance.

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