- FreeSWITCH-users - FreeSWITCH Mailing Lists

Blind transfer with call waiting
by Tiago Sampaio 03 May '13

03 May '13

Is there something like blind transfer with call waiting when the target is busy? I have a situation here, EX: When Jhon (user 1001) answer a call and and customer ask to talk to Robert (user 1002), Jhon do a blind tansfer an hangup. Currently im wasting that call when Robert is busy. I need an way to park the call and when Robert become available ring his extension (and maybe put an timeout and call back to Jhon). Im currently writing an script with perl to do this, something like that (i do this from memory): use strict; use POSIX qw(strftime); my ($target_ext,$source_ext,$tryes); my $max_time = 30; our $session; $target_ext = $session->getVariable('target_ext'); $source_ext = $session->getVariable('source_ext'); while($tryes <= $max_time){ my $uuid = $session->execute("db","select/in_a_call/$target_ext"); if(!$uuid) $session->execute("trasfer","$target_ext XML transfer_context"); sleep(1); $tryes++; } $session->execute("transfer","$source_ext XML transfer_context"); Before call an extension i do db insert/in_a_call/<EXTENSION>/${uuid} and after db delete/in_a_call/<EXTENSION> But in thinking will be too hard to maintain this code and must be an easier way to do this... Any help?

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How to retreive FreeSWITCH-IPv4 value ?
by julien terrasson 03 May '13

03 May '13

Hello, I'm trying to dump the Freeswitch-IPv4 value in one of the session. But for some reason the getvar primitive returns an "_undef_" value Should i use a different primitive to retreive that value ? Example : freeswitch@debian> uuid_dump *2df05810-b3dc-11e2-b9af-f9b7dfbfcfcb* Event-Name: CHANNEL_DATA Core-UUID: 73c3dc70-b3d4-11e2-b885-f9b7dfbfcfcb FreeSWITCH-Hostname: debian FreeSWITCH-Switchname: debian *FreeSWITCH-IPv4: 192.168.1.2* FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2013-05-03%2012%3A28%3A45 .... freeswitch@debian> uuid_getvar *2df05810-b3dc-11e2-b9af-f9b7dfbfcfcb* * FreeSWITCH-IPv4* *_undef_* Regards, Julien

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Friday Free For All
by Jeff Leung 03 May '13

03 May '13

It's Friday and you know what we do today ;)

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Check if UA is still there?
by Peter Steinbach 03 May '13

03 May '13

Hello, is there a way to check if a UA is still available? The reason is, that sometimes a channel hangs and is not being terminated because the UA was disconnected due to a network failure. As the media is outside Freeswitch (bypass media), it is difficult for us to determine, if the call is still active. Meanwhile the B leg to PSTN is also not hungup and produces costs. Is there a way Freeswitch can detect this? Or is there a way Freeswitch may send a SIP message to the UA by an external process and wait for an OK which can be captured e.g. by event socket interface? An help is welcome. -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de

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Compatibilitie between SIP V1.3.0.23 and SIP V2 Sofia Stack
by Deon Vermeulen 03 May '13

03 May '13

Hi All Due to time constraints I'm posting here for the hope for a quick response. I'm currently trying to setup a Trunk to our upstream Carrier. The issues we are facing is that inbound calls from them are working 100%. Outbound calls to them are returned with a 500 error. The only immediate differences we could pickup was the SIP versions. They have a Huawei MSOFTX3000 running SIP version 1.3.0.23. I'm running freeSWITCH 1.5.1b+git~20130414T025052Z~969eae39d9 Is there any possibility someone could help me figure out what compatibility issues between the SIP versions could cause, especially for out bound calls to the Carrier? Thank you very much Kind Regards Deon Vermeulen

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multiple same header in INVITE
by Mickael Hubert 03 May '13

03 May '13

Hi list, I have my freeswitch in SBC "mode" and topology hiding. I'd like to copy some header between A leg and B leg (not to copy all headers). My code is OK, but when I have multiple same headers in my INVITE, freeswitch only copy the last header. Ex: A Leg: Diversion: <sip:0326123456@hexanet.fr>; reason=user-busy; privacy=off Diversion: <sip:0326654321@hexanet.fr>; reason=user-busy; privacy=full ;counter=3 B leg: Diversion: <sip:0326645321@hexanet.fr >;reason=user-busy;privacy=full;counter=3 Only last Diversion header is copied. how can I copy all same header please ? my code: *<condition field="${sip_h_Diversion}" expression="(.*)" break="never">* * <action application="log" data="LOG Modifying Diversion" />* * <action application="export" data="sip_h_Diversion=$1" />* *</condition>* thanks in advance -- Cordialement Hubert Mickaël Ingénieur VOIP - Hexanet --

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XML encoding attribute ?
by Torstein Knutsen 03 May '13

03 May '13

Hi I have a questione regarding XML output in event socket or cli. Example : "+" in the fs_cli when issuing "conference xml_list" returns <caller_id_name>*%2B*4726262626</caller_id_name> When in fact my caller_id_name is "+4726262626". The printout shows no encoding attribute : <?xml version="1.0"?> I'm I correct to assume that if I can get encoding in utf-8, the "+" will be "+" and not "*%2B"* in the listing ? Does anybody know how I can accomplish this ? Im running ubuntu server 10.04 / FreeSWITCH Version 1.2.8+git~20130412T165647Z~3fde0e453e (git 3fde0e4 2013-04-12 16:56:47Z) best regards Torstein

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T38 Re-INVITE: FS responds 100 Trying, nothing further happens
by Örn Arnarson 03 May '13

03 May '13

Hello, I'm trying to get T38 gatewaying to work. I have a call coming in with FAX audio, and an endpoint that needs T38. The endpoint will send a T38 re-invite, to which FS responds 100 trying, and after 40 seconds the endpoint will give up (citing stale 100 trying). I set these variables before briding: <action application="set" data="fax_enable_t38=true"/> <action application="set" data="fax_enable_t38_request=true"/> And then I bridge like this (dialstring is sofia/profile/number@ip:port): <action application="bridge" data="{execute_on_image=t38_gateway self nocng}${dialstring}"/> I've tried all variants I can think of, and always have the same result, except for when fax_enable_t38 = false, in which case FS rejects the re-invite. I found some old discussions on the mailing list alluding to the same problem, but not much in the way of solutions. Can anybody give me a pointer as to what I'm doing wrong? I was running a git clone from 1.2.2, but I upgraded to latest stable (1.2.7) with the same results. Here's what happens in FS. At the very end you can see the Re-INVITE, and then nothing happens post that point. http://pastebin.freeswitch.org/20853 Best regards, Örn

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Copy rtmp extra header to sofia
by Iwan Budi Kusnanto 02 May '13

02 May '13

Hi, I want to copy RTMP extra header to Sofia/SIP. I can do it in dialplan like this <action application="export" data="nolocal:sip_h_X-Flash-Header1=${rtmp_u__X-Flash-Header1}"/> This way, i need to specify each header name in dialplan. Can we make dialplan that support arbitrary header name? I can do it in Lua with this method: - execute uuid_dump - search line that started by variable_rtmp_u__X-Flash - extract header_name and header_value - session:execute("export", "nolocal:sip_h_" .. header_name .. "=" .. header_value) Any better way to do this? Thanks -- Iwan Budi Kusnanto

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make ekiga to flex call
by Navnath Sonavne 02 May '13

02 May '13

Hi All, I am using flex client(at 192.168.9.165) given in freeswitch source to make call to Ekiga(h323 protocol) softphone(at 192.168.8.27) via freeswitch server(at 192.168.8.41). I have two users registered on freeswitch with extension as 1100 and 1101 in default context. I loged in using one of extension in flex client,then I dail another extension. After I dail,call goes to Ekiga softphone at 192.168.8.27 because I have made changes in default.xml accordingly to forward calls to this ip. Here is My default.xml part : <extension name="Local_Extension"> <condition field="destination_number" expression="^(\d{4})$"> <action application="export" data="dialed_extension=$1"/> <action application="set" data="effective_caller_id_number=${dialed_extension}"/> <action application="set" data="PEER_IP=192.168.8.27"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="proxy_media=true"/> <action application="bridge" data="h323/$1@${PEER_IP}"/> </condition> </extension> Ekiga phone can answer the call and there is audio transmission from both end successfully. But this is one way calling i.e. from flex to ekiga. Now I want to call from Ekiga softphone to flex client. How to call flex client from ekiga softphone? Anybody please me on this issue. -Navnath.

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