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How to detect DTMF and generate an event?
by Szeto, Steven 21 Oct '14

21 Oct '14

If I establish a call between two registered endpoints, and then press a DTMF key, I can see a DTMF key press event from the originator's call id: Freeswitch EventName: [[EVENT-NAME, DTMF]] - CallId: [694eb613-6b97-4197-a951-5288066ffbf5] () GenerateDTMFEvent - CallID: [694eb613-6b97-4197-a951-5288066ffbf5] - TimeStamp: [10/17/2014 01:03:34.993 AM] - DTMF: [5] () How can I get equivalent DTMF events on the terminator's call id? I hear the DTMF key on the phone, but I need to process the keystroke in my own application. Can the bind_digit_action application do what I need? If so, can I please get an example? -- Regards, Steve Szeto *MiContact Center IVR Team* Software Design *Phone:613-592-2122* Mobile: *Mitel* 350 Legget Drive, Kanata, Ontario, K2K 2W7, Canada mitel.com steven.szeto(a)mitel.com [image: Facebook] <http://www.facebook.com/mitel.networks>[image: Twitter] <http://www.twitter.com/Mitel>[image: Google Plus] <http://plus.google.com/+mitel>[image: Linkedin] <http://www.linkedin.com/company/mitel> -- This e-mail (including any attachments) is for the sole use of the intended recipient(s) and may contain information that is confidential and/or protected by legal privilege. Any unauthorized review, use, copy, disclosure or distribution of this e-mail is strictly prohibited. If you are not the intended recipient, please notify Mitel immediately and destroy all copies of this e-mail. Mitel does not accept any liability for breach of security, error or virus that may result from the transmission of this message.

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Re: [Freeswitch-users] Peer Identity outdated after SIP transfer
by Julio Cesar Esteves Cabezas 21 Oct '14

21 Oct '14

Let me, please, describe what I am trying to achieve: My company develops and sells a VoIP communication system and today it integrates some commercial 3rd party components that we'd like to replace by open-source components, like Freeswitch. That would make our solution more affordable to small-sized clients. We are experimenting with Freeswitch to understand which of our system requirements can be met by simply making an adequate configuration in FS and which ones will need some development. As I reported, using FS, when an UserAgent A makes an unattended transfer of UserAgent B to UserAgent C, B does not receive any signalling (Wireshark shows that) and so it cannot by any conceivable means get the update that now it is connected to C and no more to A. There is certainly an alternate way to configure FS (maybe even at the XML dialplan level) to behave differently, because any reasonably standard system requires that update to happen. So, what do you recommend in Freeswitch for that ? > Brian West brian at freeswitch.org > Sat Oct 18 17:48:47 MSD 2014 > . Previous message: [Freeswitch-users] Peer Identity outdated after SIP transfer > . Next message: [Freeswitch-users] Peer Identity outdated after SIP transfer > . Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] > ________________________________________ > that endpoint doesn't do display updates as far as I recall. > > On Fri, Oct 17, 2014 at 1:31 PM, Julio Cesar Esteves Cabezas < > jcabezas at inovax.com.br> wrote: > > > I am using version 1.5.12b~64bit ( 64bit) for Windows. > > For now I am testing with soft-phones, X-Lite e.g. > > Anyway Wireshark running in transferee X-Lite host does not show any SIP > > messages during/after the transfer process, only in the call setup and call > > teardown. > > > > Thanks. > > > > *From:* Brian West [mailto:brian at freeswitch.org] > > *Sent:* Friday, October 17, 2014 8:40 AM > > *To:* FreeSWITCH Users Help > > *Subject:* Re: [Freeswitch-users] Peer Identity outdated after SIP > > transfer > > > > > > > > This already happens on supported phones, what phones are you using? What > > FreeSWITCH rev? > > > > On Friday, October 17, 2014, <jcabezas at inovax.com.br> wrote: > > > > Hi, > > > > When user A blind-transfers user B to user C (using standard REFER > > signaling) Freeswitch does the transfer and B then talks with C, that's OK. > > But B's UA still keeps displaying A as the peer and not C, as would be > > correct. > > > > I need that information to be displayed up-to-date in the transferee UA. > > How can that be achieved ? > > - A proxy-like way of dealing with transfer signaling can be enabled in > > FS ? > > - Any specific signaling for updating peer identity, between Freeswitch > > and transferee, can be enabled in FS? > > > > Thanks, > > Julio. > > > > > >

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max-db-handles
by Guillermo Ruiz Camauer 21 Oct '14

21 Oct '14

There is a parameter in switch.conf.xml called max-db-handles. When should this be tweaked? Is there any relation to max-sessions? (example: 1 handle every XX sessions) Thanks, -- Guillermo Ruiz Camauer

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FS Installation link not found
by Mo Rajib 21 Oct '14

21 Oct '14

Never mind. Found it. Thanks.

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FS Installation link not found
by Mo Rajib 21 Oct '14

21 Oct '14

The installation link below is not found https://confluence.freeswitch.org/display/FREESWITCH/Installation Not Found The requested URL /display/FREESWITCH/Installation was not found on this server.

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FreeSWITCH.org infrastructure upgrades and changes
by Ken Rice 21 Oct '14

21 Oct '14

New Post on freeswitch.org from krice387 check it out at http://ift.tt/1vGF96v FreeSWITCH.org infrastructure upgrades and changes Beginning this today (Friday Oct 17), The FreeSWITCH Team will begin several upgrades to improve the network infrastructure that supports the project. This will include moving all the servers to a new network which will improve our network capacity and give us a bit more room to grow. It is possible that you’ll see some of the services offline as they are migrated.

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FS apt-get install fails unknown gateway
by Sean Devoy 21 Oct '14

21 Oct '14

Hi, I created a Debian box! Now I am trying to install FS. I am having server problems. http://confluence.freeswitch.org/ returns: Upgrades in Progress!!! The FreeSWITCH Team is currently Upgrading Servers and re-arranging services Please bear with us as we bring things back online! So, from http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide I did: echo 'deb http://files.freeswitch.org/repo/deb/debian/ wheezy main' >> /etc/apt/sources.list.d/freeswitch.list gpg --keyserver pool.sks-keyservers.net --recv-key D76EDC7725E010CF gpg -a --export D76EDC7725E010CF | sudo apt-key add - apt-get update apt-cache search freeswitch apt-get install freeswitch-all (also tried freeswitch-meta-vanilla) which dies with a Failed to fetch ... http://files.freeswitch.org/repo/deb/debian/pool/main/f/freeswitch/freeswit… also, from a browser: http://files.freeswitch.org/repo/deb/debian/pool/ returns 502 bad gateway. I must have missed some major planned outage. When is the dist expected back? Thanks, Sean

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Testing list
by Ken Rice 21 Oct '14

21 Oct '14

This is a list test please ignore -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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RTP issue
by Miha 21 Oct '14

21 Oct '14

Hi, first question, was any bug reported in version :FreeSWITCH Version 1.4.8+git~20140910T181155Z~9f72b11892~64bit (git 9f72b11 2014-09-10 18:11:55Z 64bit) for RTP? A lot of users are reporting that media stops in about 1:30 of the call. First I thought that is was NAT issue but now I noticed that this was not the case as this UAC's are connected via direct tunnel to Freeswitch, so this is not an issue. Problem is that in the middle of the call, exactly at 1:30 FS sends new request (INVITE) to uac, uac replay back with 200ok and media stops (sdp seems to be ok). What could be an issue, with 1.2 I did not have this issue. tnx miha

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AASTRA Phones - NAT issue
by Sean Devoy 21 Oct '14

21 Oct '14

Hi All, I am trying to get an AASTRA 480i working with FS. I have other phones working just fine (Cisco and Polycom). I have updated the firmware. I can see in the "sofia global siptrace" the REGISTER request, the Unauthorized response, but it does not appear to be making it back to the phone. That statement is based on my observation the CSEQ does not change. I have added and removed: <variable name="sip-force-contact" value="NDLB-connectile-dysfunction"/> I have tried many different option through the phones web interface, but no luck yet. Anyone have a working configuration (phone and FS) they could share)? The key siptrace packets are here (these happen to be from the attempt with both UDP and TCP): recv 811 bytes from udp/[71.121.183.4]:5060 at 19:28:05.468752: ------------------------------------------------------------------------ REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0 Via: SIP/2.0/UDP 71.121.183.4:5067;branch=z9hG4bKba321b220 Max-Forwards: 70 Content-Length: 0 To: 224 <sip:224@fs_bfis.bizfocused.com:5060> From: 224 <sip:224@fs_bfis.bizfocused.com:5060>;tag=23c56ce0efc9c8d Call-ID: 9fda297c6de44793402fc2f778f24fb0(a)71.121.183.4 CSeq: 7581459 REGISTER Contact: 224 <sip:224@71.121.183.4:5067;srcadr=10.10.40.49:5060;srcadr=10.10.40.49:5060>;expires=300 Contact: 224 <sip:224@71.121.183.4:5067;transport=tcp;srcadr=10.10.40.49:5060;srcadr=10.10.40.49:5060>;expires=300 Proxy-Require: com.nortelnetworks.firewall Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Aastra-Mac:00085D03237F Aastra-Line:1 User-Agent: Aastra 480i/1.4.0.1048 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 ------------------------------------------------------------------------ send 667 bytes to udp/[71.121.183.4]:5067 at 19:28:05.470167: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 71.121.183.4:5067;branch=z9hG4bKba321b220 From: 224 <sip:224@fs_bfis.bizfocused.com:5060>;tag=23c56ce0efc9c8d To: 224 <sip:224@fs_bfis.bizfocused.com:5060>;tag=eKgtr8tcS78QH Call-ID: 9fda297c6de44793402fc2f778f24fb0(a)71.121.183.4 CSeq: 7581459 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.22+git~20140309T212137Z~65fed130e5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="33a7d4af-e832-4d75-a284-b7002edadfc5", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 811 bytes from udp/[71.121.183.4]:5060 at 19:28:05.970923: ------------------------------------------------------------------------ REGISTER sip:fs_bfis.bizfocused.com:5060 SIP/2.0 Via: SIP/2.0/UDP 71.121.183.4:5067;branch=z9hG4bKba321b220 Max-Forwards: 70 Content-Length: 0 To: 224 <sip:224@fs_bfis.bizfocused.com:5060> From: 224 <sip:224@fs_bfis.bizfocused.com:5060>;tag=23c56ce0efc9c8d Call-ID: 9fda297c6de44793402fc2f778f24fb0(a)71.121.183.4 CSeq: 7581459 REGISTER Contact: 224 <sip:224@71.121.183.4:5067;srcadr=10.10.40.49:5060;srcadr=10.10.40.49:5060>;expires=300 Contact: 224 <sip:224@71.121.183.4:5067;transport=tcp;srcadr=10.10.40.49:5060;srcadr=10.10.40.49:5060>;expires=300 Proxy-Require: com.nortelnetworks.firewall Allow-Events: talk,hold,conference Allow:NOTIFY,REFER,OPTIONS,INVITE,ACK,CANCEL,BYE,INFO Aastra-Mac:00085D03237F Aastra-Line:1 User-Agent: Aastra 480i/1.4.0.1048 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26 ------------------------------------------------------------------------ send 667 bytes to udp/[71.121.183.4]:5067 at 19:28:05.971138: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 71.121.183.4:5067;branch=z9hG4bKba321b220 From: 224 <sip:224@fs_bfis.bizfocused.com:5060>;tag=23c56ce0efc9c8d To: 224 <sip:224@fs_bfis.bizfocused.com:5060>;tag=eKgtr8tcS78QH Call-ID: 9fda297c6de44793402fc2f778f24fb0(a)71.121.183.4 CSeq: 7581459 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.22+git~20140309T212137Z~65fed130e5~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: path, replaces WWW-Authenticate: Digest realm="fs_bfis.bizfocused.com", nonce="33a7d4af-e832-4d75-a284-b7002edadfc5", algorithm=MD5, qop="auth" Content-Length: 0 Thanks, Sean

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