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Problem witg message-waiting NOTIFY for Audiocodes ATA
by Yehavi Bourvine 10 Jun '10

10 Jun '10

Hello, Recently I've noticed that phones on our Audiocodes MP-124 adapter do not get the message waiting indicator. It seems like this behaviour has started a while ago after some upgrade (don't know whether it was Freeswitch or Audiocodes). The current versions are FS 1.0.6 and AudioCodes 5.6 >From various tests I've done with SIPSACK and tcpdumps I find the following: - after a Register command a "correct" notify is sent (i.e., the AudioCodes understands it). - later on "incorrect" notify's are sent which are rejected by the AC with "user not found". The "correct" notify is: NOTIFY sip:80633@10.64.0.8:5060 SIP/2.0 Via: SIP/2.0/UDP 10.64.0.3;rport;branch=z9hG4bKap2H5DSe6Zm7a Max-Forwards: 70 From: <sip:80633@pbx-gr.XXX>;tag=vtgH30805ZH1Q To: <sip:80633@pbx-gr.XXX>;tag=1c1710947583 Call-ID: 17109472461062010161640(a)10.64.0.8 CSeq: 131971366 NOTIFY Contact: <sip:80633@10.64.0.3:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.6-svn-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=7200 Content-Type: application/simple-message-summary Content-Length: 106 Messages-Waiting: no Message-Account: sip:80633@pbx-gr.XXX Voice-Message: 0/0 (0/0) While the "incorrect one" differs in the following two lines: From: <sip:80633@10.64.0.3 <sip%3A80633(a)10.64.0.3>>;tag=vtgH30805ZH1Q To: <sip:80633@10.64.0.3 <sip%3A80633(a)10.64.0.3>>;tag=1c1710947583 Taking the second one, changing the domain name and sending it with SIPSACK made it work correctly. (10.64.0.3 is the adderss of pbx-gr.XXX and 10.64.0.8 is the Audiocodes). Whose side is not working properly and should be asked to be fixed? Thanks! __Yehavi:

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Freeswitch in an existing phone network
by Oliver Schenk 10 Jun '10

10 Jun '10

Hi All, The company I work for currently has quite an extensive phone network which gets carried between old analogue PABXes which also has an interface to IP based phones. All the phones in our office are connected via CAT5 cable using IP, however literally hundreds of phones out in the field (we operate railway infrastructure) are on standard voice analogue phones carried through fibreoptics. Anyway, I would like to use Freeswitch purely for its IVR and TTS abilities and nothing else. So basically I just need it to act like a slave to whatever IP phone network is already out there. I'm a bit worried if I fire up freeswitch it will hijack the phone network! All our phones are accessible via a 5 digit extension. I would like Freeswitch to be behind one of those ... say 12345. If anyone within our phone network dials 12345 then Freeswitch should answer. I guess my question is...how should I go about disabling most of FreeSwitch except it's ability to pick up the phone and speak IVR/TTS and make an outgoing call via the existing phone network? Any general pointers appreciated. Thanks, Oliver Schenk

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Gateway Registration
by Samuel Macedo 10 Jun '10

10 Jun '10

Hi, I want to use the same Gateway but with different users and passwords to make and receive calls. This is the scenario: I have 2 users in a PSTN gateway, I want to connect 2 Freeswitch UserAgents (Sofia External) to this gateway so I can receive inbound calls. And to make a Outbound call I want to decide witch UserAgent I will use. Is there anyway to do this? Regards, -- Samuel Macedo Belo Horizonte - Brazil

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Call Forwarding/Diversion Dialplan
by Michael De Lorenzo 09 Jun '10

09 Jun '10

I'm trying to write a dialplan that will forward a call to a PSTN number. PSTN (Party A) Inbound call to your Freeswitch (Party B), which is forwarded out to the PSTN (Party C) I've found examples for transferring calls to other Freeswitch users, but I haven't been able to get this to work properly--where it sets a Diversion header. Can anyone help?

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Early media configuration
by Sean Holt 09 Jun '10

09 Jun '10

Hello lists, I¹m having 2 issues and wondering if someone can help. Here are my specs Most recent git from freeswitch Centos 5 I¹ve include my dialplan http://pastebin.freeswitch.org/13156 1st: When calling any endpoint that requires digit input, the endpoint does not recognize the dtmf. I¹ve done a wireshark trace and I see the dtmf output, so I know that part is working. Tech support at Bandwidth.com says I need to enable early media, so I added the ignore_early_media=true in my dialplan but still no success. I¹m thinking I don¹t have ignore early media in the right place, but I¹ve included my dialplan for viewing. 2nd: Incoming calls have no audio after the call bridges. Hard to say what the issue is, but it might have something to do with early media. Thoughts? Thanks Sean

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application=record_session leaves a "session" around?
by Robert Hadley 09 Jun '10

09 Jun '10

Using <action application="record_session" data="filename.wav"/> seems to leave a "session" around. I notice this repeated message later when I restart sofia internal profile after recording calls: 2010-06-08 15:37:56.641151 [CRIT] sofia.c:1459 Waiting for 3 session(s) (Msg repeats 13 times) The number of sessions "Waiting for" is the number of calls I have recorded. Pastebin of console debug log at: http://pastebin.freeswitch.org/13157 Is this a bug or is there something I need to do after recording a call to free the session? Thanks, Robert

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Out-Going Call Transfer Fixed
by Michael Gende 09 Jun '10

09 Jun '10

Hello, Just to follow up on my own follow-up, we've fixed our "we can't transfer an outgoing call to another local extension" problem. Turned out, the problem was in the dial plan, the local_extension extension, as one might expect. We added a line to set the local domain properly and all was well. Now, we see outbound called transfers going to, for example, 1001(a)the.real.lan.ip.of.the.device instead of 1001(a)the.wan.ip.address.Naturally, the former works and the latter don't. Hooray. Also, many thanks to the intrepid ones who asked me for more info and tried to offer some advice. Regards, Mike G.

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Sending NOTIFY message to a phone via XML-RPC
by Peter P GMX 09 Jun '10

09 Jun '10

Hello, I try to send a NOTIFY message from my application to a registered phone via XML-RPC. This is the XML which is sent: <?xml version="1.0"?> <methodCall> <methodName>freeswitch.api</methodName> <params> <param> <value> <string>sendevent</string> </value> </param> <param> <value> <string>NOTIFY,profile=internal,event-string=check-sync;reboot=false,user=200,host=192.168.178.220,content-type=application/simple-message-summary</string> </value> </param> </params> </methodCall> However I receive an error message: <?xml version="1.0" encoding="UTF-8"?>. <methodResponse>. <params>. <param><value><string>ERROR!</string></value></param>. </params>. </methodResponse>. I also tried to send the data as a struct but FS complains about needing a string as parameters. What am I doing wrong? Anybody has a valid sample XML for a NOTIFY message? Best regards Peter

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Remote Party ID
by Ken Fulmer 09 Jun '10

09 Jun '10

I've seen the following link but I'm still confused. http://wiki.freeswitch.org/wiki/Channel_Variables#sip_cid_type I realize this refers to channel variables, but is there a sip profile parameter to remove Remote Party ID? Thanks, Ken

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API pages on www.freeswitch.org
by Jan Berger 09 Jun '10

09 Jun '10

Hi, I try to access the API pages on the freeswitch site, but I have so far never been able to - they just time out? Is this just me or do others have the same problem? Jan

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