August 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Audio Problem - one way
by Erin O'Meara 18 Aug '11

18 Aug '11

I have to VOIP servers in the cloud(same provider), one is FusionPBX running Freeswitch the other is Elastix with Asterisk. I have two phones(same Grandstream GXP280) in my office each registered to one of the servers. I have one phone number and when I route it thru the Elastix server, I get no complaints about the audio. When I route the number thru the FusionPBX the person on the other end complains of choppy audio but I can't tell the difference on my end. But when I call from my cell I can tell that the Freeswitch server audio is not as good on the outgoing. Both Virtual servers are the same specs, memory and proc. The elastix box has more overhead with all the extra crap it runs and I can look at the performance metric's on the virtual servers and confirm that the fusionpbx has less proc usage and memory usage. The calls are using the same codec (ulaw). Currently the call volume is just my one call for the Fusion server, and at most 2 for the Elastix. Any thoughts on with the same setup and internet routes, audio in one direction is preforming so poorly with the freeswitch? Regards, 206.905.9520 http://salmonbaytechnology.com

1 0

voicemail is not saved
by Yungwei Chen 18 Aug '11

18 Aug '11

Hi, I left several voicemails (Each is longer than 3 sec) to a user account, but none is available when I check the mailbox. Relevant settings are listed below. What am I missing here? Thanks. In conf/autoload_configs/modules.conf.xml, mod_voicemail is already loaded. freeswitch@internal> load mod_voicemail +OK Reloading XML -ERR [Module already loaded] freeswitch@internal> 2011-08-15 14:32:10.666978 [WARNING] switch_loadable_module.c:998 Module mod_voicemail Already Loaded! Here's the content of conf/autoload_configs/voicemail.conf.xml: <configuration name="voicemail.conf" description="Voicemail"> <settings> </settings> <profiles> <profile name="default"> <param name="file-extension" value="wav"/> <param name="terminator-key" value="#"/> <param name="max-login-attempts" value="3"/> <param name="digit-timeout" value="10000"/> <param name="min-record-len" value="3"/> <param name="max-record-len" value="300"/> <param name="max-retries" value="3"/> <param name="tone-spec" value="%(1000, 0, 640)"/> <param name="callback-dialplan" value="XML"/> <param name="callback-context" value="default"/> <param name="play-new-messages-key" value="1"/> <param name="play-saved-messages-key" value="2"/> <param name="login-keys" value="0"/> <param name="main-menu-key" value="0"/> <param name="config-menu-key" value="5"/> <param name="record-greeting-key" value="1"/> <param name="choose-greeting-key" value="2"/> <param name="change-pass-key" value="6"/> <param name="record-name-key" value="3"/> <param name="record-file-key" value="3"/> <param name="listen-file-key" value="1"/> <param name="save-file-key" value="2"/> <param name="delete-file-key" value="7"/> <param name="undelete-file-key" value="8"/> <param name="email-key" value="4"/> <param name="pause-key" value="0"/> <param name="restart-key" value="1"/> <param name="ff-key" value="6"/> <param name="rew-key" value="4"/> <param name="skip-greet-key" value="#"/> <param name="previous-message-key" value="1"/> <param name="next-message-key" value="3"/> <param name="skip-info-key" value="*"/> <param name="repeat-message-key" value="0"/> <param name="record-silence-threshold" value="200"/> <param name="record-silence-hits" value="2"/> <param name="web-template-file" value="web-vm.tpl"/>    <param name="operator-extension" value="operator XML default"/> <param name="operator-key" value="9"/> <param name="vmain-extension" value="vmain XML default"/> <param name="vmain-key" value="*"/>   <email> <param name="template-file" value="voicemail.tpl"/> <param name="notify-template-file" value="notify-voicemail.tpl"/>  <param name="date-fmt" value="%A, %B %d %Y, %I %M %p"/> <param name="email-from" value="${voicemail_account}@${voicemail_domain}"/> </email>      </profile> </profiles> </configuration> In conf/directory/default.xml, user 91000 is defined in domain voicemail_2. <domain name="voicemail_2"> <groups> <group name="default"> <users> <user bluebox_id="2" id="91000"> <params> <param name="sip-forbid-register" value="true"/> <param name="vm-password" value="xxxx"/> <param name="vm-message-ext" value="wav"/> <param name="vm-mailto" value="xxx(a)xxx.com"/> <param name="email-addr" value="xxx(a)xxx.com"/> <param name="vm-delete-file" value="false"/> <param name="vm-attach-file" value="true"/> <param name="vm-email-all-messages" value="true"/> </params> <variables> <variable name="timezone" value="America/Chicago"/> </variables> </user> </users> </group> </groups> </domain> In my dialplan, calls to 1112223333 will be sent to user 91000's voicemail box if they are not answered. <extension name="main_number_2" continue="true"> <condition field="destination_number" expression="^1112223333$"> <action application="set" bluebox="settingEndBridge" data="hangup_after_bridge=true"/> <action application="set" bluebox="settingFail" data="continue_on_fail=true"/> <action application="set" bluebox="autoPlay" data="vm_auto_play=false"/> <action application="set" bluebox="settingTimeout" data="call_timeout=30"/> <action application="set" bluebox="settingRing" data="ringback=${us-ring}"/> <action application="set" bluebox="settingRingXfer" data="transfer_ringback=${us-ring}"/> <action application="export" bluebox="sipCalleeIdName" data="sip_callee_id_name=m1"/> <action application="export" bluebox="sipCalleeIdNumber" data="sip_callee_id_number=1112223333"/> <action application="bridge" data="user/91000@$${location_3}"/> <action application="answer"/> <action application="voicemail" data="default voicemail_2 91000"/> <action application="hangup"/> </condition> </extension>

3 4

Original Caller ID/number after attended transfer
by Weigel, Stefan 18 Aug '11

18 Aug '11

0 *H÷ 010 +0 *H÷ $ Content-Type: multipart/alternative; boundary="----=_NextPart_000_0C65_01CC5DB3.B4FD4C90" Dies ist eine mehrteilige Nachricht im MIME-Format. ------=_NextPart_000_0C65_01CC5DB3.B4FD4C90 Content-Type: text/plain; charset="us-ascii" Content-Transfer-Encoding: 7bit Hi all, is there a possibility to display the original caller ID & number after doing a attended transfer. External call to phone A -> calls phone B (I see caller ID & number of phone A) -> doing an attended transfer of external call to phone B (still caller ID & number of phone A). Thanks in advance and best regards, Stefan ------=_NextPart_000_0C65_01CC5DB3.B4FD4C90 Content-Type: text/html; charset="us-ascii" Content-Transfer-Encoding: quoted-printable m<html xmlns:v=3D"urn:schemas-microsoft-com:vml" = xmlns:o=3D"urn:schemas-microsoft-com:office:office" = xmlns:w=3D"urn:schemas-microsoft-com:office:word" = xmlns:x=3D"urn:schemas-microsoft-com:office:excel" = xmlns:p=3D"urn:schemas-microsoft-com:office:powerpoint" = xmlns:a=3D"urn:schemas-microsoft-com:office:access" = xmlns:dt=3D"uuid:C2F41010-65B3-11d1-A29F-00AA00C14882" = xmlns:s=3D"uuid:BDC6E3F0-6DA3-11d1-A2A3-00AA00C14882" = xmlns:rs=3D"urn:schemas-microsoft-com:rowset" xmlns:z=3D"#RowsetSchema" = xmlns:b=3D"urn:schemas-microsoft-com:office:publisher" = xmlns:ss=3D"urn:schemas-microsoft-com:office:spreadsheet" = xmlns:c=3D"urn:schemas-microsoft-com:office:component:spreadsheet" = xmlns:odc=3D"urn:schemas-microsoft-com:office:odc" = xmlns:oa=3D"urn:schemas-microsoft-com:office:activation" = xmlns:html=3D"http://www.w3.org/TR/REC-html40" = xmlns:q=3D"http://schemas.xmlsoap.org/soap/envelope/" = xmlns:rtc=3D"http://microsoft.com/officenet/conferencing" = xmlns:D=3D"DAV:" xmlns:Repl=3D"http://schemas.microsoft.com/repl/" = xmlns:mt=3D"http://schemas.microsoft.com/sharepoint/soap/meetings/" = xmlns:x2=3D"http://schemas.microsoft.com/office/excel/2003/xml" = xmlns:ppda=3D"http://www.passport.com/NameSpace.xsd" = xmlns:ois=3D"http://schemas.microsoft.com/sharepoint/soap/ois/" = xmlns:dir=3D"http://schemas.microsoft.com/sharepoint/soap/directory/" = xmlns:ds=3D"http://www.w3.org/2000/09/xmldsig#" = xmlns:dsp=3D"http://schemas.microsoft.com/sharepoint/dsp" = xmlns:udc=3D"http://schemas.microsoft.com/data/udc" = xmlns:xsd=3D"http://www.w3.org/2001/XMLSchema" = xmlns:sub=3D"http://schemas.microsoft.com/sharepoint/soap/2002/1/alerts/"= xmlns:ec=3D"http://www.w3.org/2001/04/xmlenc#" = xmlns:sp=3D"http://schemas.microsoft.com/sharepoint/" = xmlns:sps=3D"http://schemas.microsoft.com/sharepoint/soap/" = xmlns:xsi=3D"http://www.w3.org/2001/XMLSchema-instance" = xmlns:udcs=3D"http://schemas.microsoft.com/data/udc/soap" = xmlns:udcxf=3D"http://schemas.microsoft.com/data/udc/xmlfile" = xmlns:udcp2p=3D"http://schemas.microsoft.com/data/udc/parttopart" = xmlns:wf=3D"http://schemas.microsoft.com/sharepoint/soap/workflow/" = xmlns:dsss=3D"http://schemas.microsoft.com/office/2006/digsig-setup" = xmlns:dssi=3D"http://schemas.microsoft.com/office/2006/digsig" = xmlns:mdssi=3D"http://schemas.openxmlformats.org/package/2006/digital-sig= nature" = xmlns:mver=3D"http://schemas.openxmlformats.org/markup-compatibility/2006= " xmlns:m=3D"http://schemas.microsoft.com/office/2004/12/omml" = xmlns:mrels=3D"http://schemas.openxmlformats.org/package/2006/relationshi= ps" xmlns:spwp=3D"http://microsoft.com/sharepoint/webpartpages" = xmlns:ex12t=3D"http://schemas.microsoft.com/exchange/services/2006/types"= = xmlns:ex12m=3D"http://schemas.microsoft.com/exchange/services/2006/messag= es" = xmlns:pptsl=3D"http://schemas.microsoft.com/sharepoint/soap/SlideLibrary/= " = xmlns:spsl=3D"http://microsoft.com/webservices/SharePointPortalServer/Pub= lishedLinksService" xmlns:Z=3D"urn:schemas-microsoft-com:" = xmlns:st=3D"" xmlns=3D"http://www.w3.org/TR/REC-html40"> <head> <META HTTP-EQUIV=3D"Content-Type" CONTENT=3D"text/html; = charset=3Dus-ascii"> <meta name=3DGenerator content=3D"Microsoft Word 12 (filtered medium)"> <style>  </style>  </head> <body lang=3DDE link=3Dblue vlink=3Dpurple> <div class=3DSection1> Hi all,<o:p></o:p> <o:p> </o:p> is there a possibility to = display the original caller ID & number after doing a attended transfer. = External call to phone A -> calls phone B (I see caller ID & number of phone A) = -> doing an attended transfer of external call to phone B (still caller ID & = number of phone A).<o:p></o:p> <o:p> </o:p> <o:p> </o:p> Thanks in advance and best = regards,<o:p></o:p> <o:p> </o:p> Stefan<o:p></o:p> </div> </body> </html> 1 ------=_NextPart_000_0C65_01CC5DB3.B4FD4C90-- 20é0Ñ £0 *H÷ 0>10 UDE10U Allianz Group10UAllianz UserCA0 100825090218Z 120826090218Z0j10 UDE10U Allianz110/ *H÷ "stefan.weigel(a)allianz-warranty.com10U Stefan Weigel0"0 *H÷ 0 ^umÉ/MOõ88)AfN8÷ÒpD,Æí ÃI-Ra±«÷êÕJ8ðäðÞ\RI.Doo4Ê8Ð7#{ÃL8.}D¢0¬µÊ®íà~¥L172ÔùPUê©Áv×Ð2"2;ºd€xt/®0Yï8¶Fd^Æ<wÝCbÌgÊ[üOd^€B2#âõß¬{.ÁØÅT|¯;éY1 Î#¥hnñéÜ*uz©vÛIöG&aú>±ÅÛèŒ¿S¶:]âê³€qÜòí-¹_=¢^LÁûë-*Y£Ã0À0Uÿ00U#0ÞÄÖÿzHÀs¿þo²ôõøl0U%0 +0-U&0$"stefan.weigel(a)allianz-warranty.com0IU B0@0> +·w000.+"https://rootca.allianz.com/userca/0 *H÷ Æ±G-?Áé 0[kÖ1Ìõ0Æ9 ÑrG¯&sk VïpcìþÀ:íÅùÎÀ`ôGÜ|Êñœ;Ø3ºÞÈêIòó*ÁœjdQC8ËÎsª_Ââ®u&Yr¢»ÎV4' Oê©êµ¬¡×X÷ßn>±äÍ @Oþ?<zÿ %oÐTÌÂW~ 72I8[³ÕVMöTbÙc±éÍ³h³ŠØRNq*Ü÷÷úêºînçßºú`³]^èYEl,Ûš°¶oXr>¶ÑRH$bËþ:%tôµÑŽs@ëÝû0,0 ±l0 *H÷ 0>10 UDE10U Allianz Group10UAllianz UserCA0 110302152445Z 130303152445Z0j10 UDE10U Allianz110/ *H÷ "stefan.weigel(a)allianz-warranty.com10U Stefan Weigel0"0 *H÷ 0 ŠÄãZù«V€Xð,Êd©dP²b¹Ux^ôiDª#î³z0éarm e²2QšcC,¡ëó€}ÿåÊhWÌ÷~cPØ&ÚlÕõÒSv"µUrŠÏx^HÞèy[M±fòFÛ^j,·ñUÂ³_co;5O¬{§PÑ³Þ ØèJŒa.ŸÅÎÕ®ÕKÔ9ñËäÞÄÈoETbÔuütNg«,è7ÖæTý&µeâöïüªÀú²RxM§4tùîS¥ÉC»Ô$¹"3áLfÞEfNÒ¿$¡)øß£00Uÿ0U#0ÞÄÖÿzHÀs¿þo²ôõøl0U%0++05U.0,0* ( &$http://rootca.allianz.com/userca.crl0IU B0@0> +·w000.+"https://rootca.allianz.com/userca/0-U&0$"stefan.weigel(a)allianz-warranty.com0 *H÷ [ &þ\pQ$:HÙ,t^ßyùònó_A jCÔÀÜqV,oú; 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1 0

Freeswitch ignoring bye
by Simon Leck 18 Aug '11

18 Aug '11

Hi Everyone, Does anybody does why freeswitch is not acknowledging or ignoring BYE message and if so how do we fix it? Thanks in advanced to anyone who can tell me how to fix this issue. Thanks again Simon

2 1

Dynamically switching DTMF mode?
by Dmitry Sytchev 18 Aug '11

18 Aug '11

Hi all I'm trying to achieve dynamic DTMF mode on incoming/outgoing calls. I have three profiles - one for outgoing/incoming calls to/from gateways, one for local registrations, and one for transferred call rerouting from bind_meta_app att_xfer. I faced some strange problems when tried to maintain inband DTMF path for call through FS. How should I set initial DTMF modes on profiles to allow changing DTMF mode on inbound calls? Is it enough to set dtmf_type to 'inband' and originate call? On which leg of call I should do this? And what to do with calls going through several profiles? As far as I understand, I need late negotiation turned on to disable automatic RFC2833 negotiation on inbound calls? -- Best regards, Dmitry Sytchev, IT Engineer

2 2

Multiple Registration
by DJB International 18 Aug '11

18 Aug '11

I have some question regarding the multiple registration. When I registered 2 devices, even though both of them got registered and I can make calls from both devices, but when I list, I can only see the last device that got registered on the list from sofia status profile internal. However, when I comment out the multiple-registrations value=true in sofia profile, I can then see 2 devices listed. Questions: 1) Is that normal behavior that I need to make multiple-registrations value=true before I can see 2 devices listed when I run sofia status profile internal ? 2) How come I can only see one device listed even though both devices got registered in the first situation? Thank you, Dorn B.

3 3

Re: [Freeswitch-users] [Freeswitch-dev] Meetup Paris?
by Giovanni Maruzzelli 18 Aug '11

18 Aug '11

I have fond memories of a conference I gave couple of years ago in Hyderabad and the nice reception has had. Certainly there are lot of people and companies that can be interested in FreeSWITCH, and Cluecon India. Most people is still on Asterisk, but I believe they'll be very interested into knowing first hand what are the advantages of using FS, maybe in powerful combination with OpenSips. Let's have the ball rolling, try to gauge what interest you can gather, and which companies can act as sponsors (money matters ;) ). -giovanni On Tue, Aug 16, 2011 at 8:28 AM, Prashant Lamba <prashant.lamba(a)gmail.com> wrote: > Colleagues, > After seeing the success of ClueCon, I have always been keen to promote open > source telephony here in India and possibly organize an "India" leg of > something like ClueCon here in Hyderabad. I can start the ground work of > gathering people, speakers, venue etc. > Hyderabad is probably the best technology hub here in India. Any > suggestions? > Prashant > -- > Prashant Lamba > Phonologies (India) Private Limited > e: prashant(a)phonologies.com > m: +91 9867.22.1975 > > On Fri, Aug 12, 2011 at 10:58 AM, Mathieu Rene <mrene_lists(a)avgs.ca> wrote: >> >> English version follows. >> >> En cette fin de ClueCon, je pars sur Paris demain soir (donc j'arrive >> samedi en matiné, bien décalé), et j'y resterai jusqu'à mercredi. Je >> propose d'organiser un petit meetup afin de joindre l'utile à l'agréable et >> de voir un peu le visage de la communauté freeswitch parisienne. Est-ce >> quelqu'un peut proposer un bar sympa ou on pourrait faire un truc? >> >> --- >> >> ClueCon is over and I'm leaving to Paris tomorrow (landing saturday >> morning, well jetlagged) and staying until Wednesday. I'm proposing to >> organize a little meetup in order to get to know the face of the parisian >> freeswitch community. Does anyone know a bar where we could meet up? >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene(a)avgs.ca >> >> >> >> >> >> _______________________________________________ >> Join us at ClueCon 2011, Aug 9-11, Chicago >> http://www.cluecon.com 877-7-4ACLUE >> >> FreeSWITCH-dev mailing list >> FreeSWITCH-dev(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> http://www.freeswitch.org > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-dev mailing list > FreeSWITCH-dev(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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Re: [Freeswitch-users] SIP proxy collect DTMF using FS
by Sam Govind 18 Aug '11

18 Aug '11

Thanks for taking your time Gohar, I've looked at the link and this indeed seems like helping. Brain, can you suggest any other possibilities for this? I'd definitely like to go for the simple solution. On Wed, Aug 17, 2011 at 9:54 AM, Gohar Ahmed <gohar.ahmed(a)vopium.com> wrote: > Hey, **** > > ** ** > > Don’t know if this will do the job or not but do take a look at deflect > application: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect > **** > > This will remove the FS from the call path and tell the originator to go > somewhere else. **** > > ** ** > > I hope this helps, maybe some other gurus here suggest you anything better > than this.**** > > ** ** > > Regards,**** > > Gohar A.**** > > ** ** > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Sam Govind > *Sent:* Monday, August 15, 2011 10:58 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] SIP proxy collect DTMF using FS**** > > ** ** > > Thanks Brian for showing concern. I'm always open for ideas. What I'm > trying to achieve is collect DTMF from user and then have my SIP Proxy > verify if a particular caller is allowed to dial that destination(input as > DTMF).**** > > ** ** > > Obviously I could've done the same checks at FS, BUT I'm *required *to let > SIP Proxy verify instead. FreeSWITCH is only required to get input, release > the call and if Proxy allows the call only then call be routed to any other > FreeSWITCH (Pool of FS LoadBalanced).**** > > ** ** > > This is supposed to simplify the operations of FS , decrease the load > volume on FS, and increase the call capacity.**** > > ** ** > > If you've any better Ideas do share.**** > > ** ** > > On Sun, Aug 14, 2011 at 8:39 AM, Brian West <brian(a)freeswitch.org> wrote:* > *** > > What are you trying to do exactly because it sounds like you have selected > the most painful way to accomplish it?**** > > ** ** > > /b**** > > ** ** > > On Aug 9, 2011, at 2:11 AM, Sam Govind wrote:**** > > > > **** > > Hi guys, > > I'm looking to establish a scenario like this, any idea how to do it, if > its > possible. > > 1- SIP proxy send call to FS where DTMF will be collected. (I'm thinking of > using PlayAndGetDigits) > 2- DTMF collected be sent back to SIP proxy while FS ends the call > 3- Call at SIP proxy end keeps running for some other processing. > > basically I just need FS to collect DTMF and send those back to SIP Proxy. > > Any ideas are welcome. > > Thanks.**** > > ** ** > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > 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**** > > ** ** > > > 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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Bad port when using rport
by Spencer Thomason 17 Aug '11

17 Aug '11

Hello all, Our FS server is on a public IP on a dedicated server at one location and all clients are behind nat or vpn from other locations. We have a few "road warrior" clients when our employees travel. I'm seeing the following from one of the client devices. I'm not sure of any of the details of the router (or have anyway to control it) but for some reason the rport is sent with an invalid port number. Notice the 8 at the end of the port number is duplicated. I'm seeing the same behavior from both a Bria client and Zoiper. Is there any way to work around this? An excerpt from Wireshark: 466 2011-08-17 11:36:04.432912 80.34.4.95 206.125.40.171 SIP Request: REGISTER sip:x.x.x.x Internet Protocol, Src: 80.34.4.95 (80.34.4.95), Dst: 206.125.40.171 (x.x.x.x) User Datagram Protocol, Src Port: 56238 (56238), Dst Port: sip (5060) Session Initiation Protocol Request-Line: REGISTER sip:x.x.x.x SIP/2.0 Message Header Via: SIP/2.0/UDP 80.34.4.95:562388;rport;branch=z9hG4bKPjSAbzV32ywqJ8xWjWLx81Ml6oQcNlrM.s Max-Forwards: 70 From: "XXXX" <sip:1001@x.x.x.x>;tag=rPFQBJFIKAQpE88mHeiZ7ic59suQY70w To: "XXXX" <sip:1001@x.x.x.x> Call-ID: awr-KAMoJpMYEt2mqkOkknkRrCXUj.Jv CSeq: 27137 REGISTER User-Agent: Bria iPhone 1.2.12 Contact: "XXXX" <sip:1001@80.34.4.95:562388> Expires: 900 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0

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Inband DTMF when rfc2833 negotiated?
by Avi Marcus 17 Aug '11

17 Aug '11

Why is this call using inband when the SDP says rfc2833? I got complaints that the DTMF wasn't working, despite the dtmf numbers in the log being correct. Trace on leg B: http://pastebin.freeswitch.org/17053 Thanks, -Avi p.s. I think I've seen this with toll free gateway, too.

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