May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Aastra and SCA
by Michael Jerris 04 May '10

04 May '10

Aastra has a broken SCA implementation. We have been working with them to resolve it but as of yet they have failed to deliver a working firmware for their phones. Mike On Apr 30, 2010, at 10:07 PM, Mark Maly wrote: > Hi, > > I’ve tried to patiently figure this out by reading the wiki and this list. Unfortunately, I’ve been unable to get it right. > > I have 2 Aastra 6731is and a 51i and trying to get SCA working. I’m experiencing problems similar to the Cisco thread from last month – outgoing calls implement SCA well. Incoming calls ring all lines and appearances work, but when one phone is answered, the line appearances are removed from the remaining phones. I am not attempting to use any DNS. > > My configuration has all three phones plus FS on a local LAN. Nothing too fancy. Each line is configured for Broadsoft SCA and SCA bridging is enabled globally for the phones. > > Trying to update/replace an old phone at my church.

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Trouble With Transfer and Async Event Socket
by Joe Crivello 03 May '10

03 May '10

Hello All, I am currently working on a conference bridge based on FreeSWITCH that allows some conference members to dial out to a prompted number. Currently my implementation binds * in conference.xml to the "transfer" action, leading to a dial plan extension that runs an async full outgoing event socket. My application serving the event socket does the following: 1) Verifies the user is allowed to dial out, and writes a record to the database. 2) Retrieves the number to be dialed with 'play_and_get_digits' 3) Bridges the call to the dialed number 4) Waits for the a-leg to press star 5) Uses 'transfer' to transfer both legs back to the conference Everything works up to step #5, which does nothing. There is no NOTICE outputted to the console indicating that a transfer is taking place and neither leg is rejoined to the conference. I get a normal event back after the transfer sendmsg: Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: %2BOK If I subscribe to all events before running the transfer and receive events in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events after the transfer. I am using ESL to communicate with FreeSWITCH, and prior to running any of steps #1-5 I set the event lock to true. The line from the dial plan extension that calls my application follows: <action application="socket" data="172.18.11.1:8001 async full" /> Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS. Thanks! Joe Crivello

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Cisco 7960 has problem with music on hold
by Gary Chen 03 May '10

03 May '10

I just installed FS and registered two cisco 7960 with default configuration file. They can talk to each other. After connecting both phones, I can put one on hold with music, but I can not get the connection back by pressing resume softkey button. Once I pressing resume button, the connection was hangup. I can see the cisco 7960 sent INVITE to freeswith once I press resume button but FS sent back 'SIP/2.0 500 Overlapping Requests ' message. This cause cisco 7960 sent out BYE message. Any idea? Gary

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Anyone have analog Fax working through FS and Sangoma cards?
by Robert Hadley 03 May '10

03 May '10

Does anyone have analog Fax working through Freeswitch and Sangoma A101 & A200/FXS cards? If so, would you share what configuration and dialplan settings need to be made? I have a problem receiving Faxes, which only work 80% of the time and often take 10-12 minutes per Fax, versus 2 minutes when using the Fax machine on analog POTS line. My setup: SendingFax --- {PSTN} --- PRI --- A101 --- FS --- A200/FXS --- MyFax Pastebin to dialplan, wanpipe conf, vars.xml, internal.xml: http://pastebin.freeswitch.org/12884 Any help would be greatly appreciated. Thanks, Robert

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Trouble With 'transfer' DP After a 'bridge' on Async Event Socket
by Joseph Crivello 03 May '10

03 May '10

Hello All, I am currently working on a conference bridge based on FreeSWITCH that allows some conference members to dial out to a prompted number. Currently my implementation binds * in conference.xml to the "transfer" action, leading to a dial plan extension that runs an async full outgoing event socket. My application serving the event socket does the following: 1) Verifies the user is allowed to dial out, and writes a record to the database. 2) Retrieves the number to be dialed with 'play_and_get_digits' 3) Bridges the call to the dialed number 4) Waits for the a-leg to press star 5) Uses 'transfer' to transfer both legs back to the conference Everything works up to step #5, which does nothing. There is no NOTICE outputted to the console indicating that a transfer is taking place and neither leg is rejoined to the conference. I get a normal event back after the transfer sendmsg: Event-Name: SOCKET_DATA Content-Type: command/reply Reply-Text: %2BOK If I subscribe to all events before running the transfer and receive events in a loop, I receive nothing but routine RE_SCHEDULE and CALL_UPDATE events after the transfer. I am using ESL to communicate with FreeSWITCH, and prior to running any of steps #1-5 I set the event lock to true. The line from the dial plan extension that calls my application follows: <action application="socket" data="172.18.11.1:8001 async full" /> Any ideas? Perhaps I'm "doing it wrong"? I am very new to FS. Thanks! Joe Crivello

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Need help on realtime integration Freeswitch with Opensips.
by hung nguyen 03 May '10

03 May '10

Hi list. I have deployed cluster FS with share DB - using ODBC in core. I think it is great if we can deploy realtime integration Freeswitch with Opensips. With this, Opensips will act as register,proxy server and load balancer in front of FS boxs. Anybody have same ideal and done this. Tks.

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Strange errors from sqlite DB
by Peter Olsson 03 May '10

03 May '10

I've had an instance of FS running on a box for a couple weeks (since I last touched it), it's mostly used for trunking between some lab PBX'es and conferencing, and 2-3 Polycom's also register to it. It was compiled at april 14 from current git at that time (I can look up the exact revision if needed). Today, after a restart, I suddenly received these errors. I've never seen them before, so I was quite surprised :) Has anyone seen these before? I shut down FS after the error, removed all db-files and restarted again. The error dissapeared, but is this something I should worry about? I've been using FS (very lightly - but still..) for over a year, and this was the first time I noticed it. I found on the list that someone had a similar problem before, but I couldn't find a real explanation to it. I'm running on Windows 2003. ---- 2010-05-03 20:29:21.032501 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence] insert into sip_authentication (nonce,expires,profile_name,hostname, last_nc) values('6e5df620-d388-4f93-b0b0-8da4cf880ef1', 1272911421, '(NULL)', 'internal', 0) 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:722 SQL ERR: [select nonce,last_nc from sip_authentication where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1' and last_nc < 1] library routine called out of sequence 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence] update sip_authentication set expires='1272911421',last_nc=1 where nonce='6e5df620-d388-4f93-b0b0-8da4cf880ef1' 2010-05-03 20:29:21.048126 [ERR] switch_core_sqldb.c:404 SQL ERR [library routine called out of sequence] Regards, Peter

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mobile phone clients / fs cluster
by Phillip Jones 03 May '10

03 May '10

Hi there, I have two FS servers (FS1 and FS2) behind an inbound OpenSIPS proxy. Outbound (terminating) traffic goes directly from each FS box. All my home office phones get calls no matter which box they are registered with or which box the call comes in on - thanks Anthony. However my SIP client on various iPhone/Androids etc only receive calls that originate on the box on which they are registered. Looking at the SIP trace - when the call comes in on the 'wrong' box, the invites to these SIP clients do not even get a response. Presumably because the 3G network has no idea who this new IP in the "from address" is, who trying to contact them. Question is, is there a way around this - our will I have start routing all terminating traffic out through the proxy also. Any insight appreciated. Thanks! Pj

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How to send "waiting call" tone to remote side
by Yehavi Bourvine 03 May '10

03 May '10

Hello, At the default dialplan of freeswitch, when there is a waiting call the remote side hears a normal "ring" tone. I would like to change it to send a "call waiting" ring tone. What is the official way of doing so? I can think of two ways: - sending 182 instead of 180. - send 180 with early media and play the ring tone. I've tried the second method. It works inside our PBX and at the adjacent telco PBX. On other telco PBXes it doesn't work (silence). Before I call the telco I would like to know what is the common way of doing so. Thanks! __yehavi:

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absolute_codec_string vs codec_string
by Ken Fulmer 03 May '10

03 May '10

Would someone please clarify the difference between the following parameters? I've seen the Wiki page but still don't understand the difference. 1. absolute_codec_string 2. codec_string Thanks, Ken Fulmer

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