June 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Fwd: RECURSIVE ERROR
by Rodrigo Pinto 05 Jun '13

05 Jun '13

Hello, We recently migrated to latest stable version 1.2.9. And in the console this error is appearing randomly: 2013-06-05 18:14:00.089079 [ERR] switch_ivr.c:144 RECURSION ERROR! It's not the best idea to call things that collect input recursively from an input callback. I see that this version has this check in ./include/switch_types.h: #define MAX_ARG_RECURSION 25 #define arg_recursion_check_start(_args) if (_args) { \ if (_args->loops >= MAX_ARG_RECURSION) { \ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, \ "RECURSION ERROR! It's not the best idea to call things that collect input recursively from an input callback.\n"); \ return SWITCH_STATUS_GENERR; \ } else {_args->loops++;} \ } Any idea how to easily increase MAX_ARG_RECURSION or disable it like older versions? Thanks!

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FreeSWITCH Conference Call Today
by Michael Collins 05 Jun '13

05 Jun '13

Hello all! Don't forget about the community conference call today. The agenda is very light today so the audience will be having lots of input. Here is our agenda page for today: http://wiki.freeswitch.org/wiki/FS_weekly_2013_06_05 Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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how to do callback with dialplan?
by Vincent Xia 05 Jun '13

05 Jun '13

how to do callback with dialplan? the idea is when user A(his phone number is, for example 1001) calls a number with a prefix, e.g. **1002, the freeswitch will hangup the call leg with user A and originate calls to 1001 and 1002, then bridge 1001 and 1002 after both are connected. since originate is a console command, how could i make dialplan to do callback? maybe it's something like: <extension name="callback"> <condition field="destination_number" expression="^\*\*(\d+)$"> <action application="hangup"/> <action application="originate" data="sofia/${use_profile}/$1"/> <action application="originate" data="sofia/${use_profile}/${caller_id_number}"/> <action application="bridge" data="sofia/${use_profile}/$1 sofia/${use_profile}/${caller_id_number}"/> </condition> </extension> but certainly it's not gonna work, can anybody give a hand?

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client doesn't respond to picture fast update message delays BYE to be sent
by Seven Du 05 Jun '13

05 Jun '13

Hi there, I have FS call a doubango based video client and the client returns 180 and 200 with the following allow header. Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE But when FS send picture_fast_update message in INFO doubando completely ignore it and doesn't reply anything that results more INFO resend and, FS won't send BYE immediately when hangup the channel but delays about 30 seconds, which I guess some state machine still waiting for the INFO response? so question: 1) Because it doesn't "Allow" INFO, so is it right to just ignore the INFO when we sent to it anyway or should reply an 4xx or other error? 2) Should we avoid to send INFO because the client doesn't support? Is there anywhere to disable it or should I report a jira? FS version is: User-Agent: FreeSWITCH-mod_sofia/1.5.2b+git~20130604T221005Z~b4ed2f55bf Thanks. -- Seven Du http://www.freeswitch.org.cn http://about.me/dujinfang http://www.dujinfang.com Sent with Sparrow (http://www.sparrowmailapp.com/?sig)

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How can I tell sangoma card to do not transcode
by Mino Haluz 05 Jun '13

05 Jun '13

Hi, I'm doing a research regarding transcoding before buying sangoma card, I have two questions guys: 1) If I use card with 240 sessions, it means I can transcode just 120 CC ( http://wiki.sangoma.com/FreeSWITCH-transcoding) right? 2) Can I tell FS to use Sangoma card only if transcoding with G729 ? Because it seems to me that SW transcoding is cheaper than HW transcoding when dealing with G711/G722/GSM, isnt it? As well as 20$ per 1 CC G729 license for SW transcoding. Thanks, Mino

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Trouble with git
by Terry Ray 05 Jun '13

05 Jun '13

I have an existing freeswitch install which I'm trying to upgrade. I attempted to run the: make current but it bombed after the cleans with: fatal: unable to connect to git.freeswitch.org: git.freeswitch.org[0: 198:22.64.222]: errno=Connection refused. So, I just tried a git pull, same result. I can connect to other git sites (example github), just not the freeswitch git repos. Is it down or has it changed locations? Please advise. Thanks, T

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Standards support in FreeSwitch
by James Lotery 05 Jun '13

05 Jun '13

Hello, we are using FS to interconnect with a large carrier and as part of the process the carrier has given us a list of RFCs and ITUs that we need to confirm whether or not they are supported by our "equipment". Rather than post the list here is there a resource available somewhere that I can use to see what standards are supported or implemented in FS? I've searched the wiki on the FS web site and didn't find anything exhaustive. Thanks for any help in advance. James

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Is it possbile to remove Allow, Supported, Allow-Events from header in freeswitch
by divyeshkamothi 05 Jun '13

05 Jun '13

hi all there is requirement where i need to remove Allow, Supported, Allow-Events. from fs SIP headers. please provide me way using which i can remove. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Is-it-possbile-to-remove-Allo… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Issue with changing phone SIP port
by Oleg Stolyar 05 Jun '13

05 Jun '13

Hi guys, I ran into the following problem recently: Using a softphone from a computer behind a NAT I register it with FreeSWITCH. It registers with a certain port in the Contact header. FreeSWITCH stores this port in the user's registration info and uses it from then on to send messages to the phone. However, for some reason FreeSWITCH initially sends back an "Unauthorized" response. After that some phones seem to send REGISTER again but with a*different port *. This only happen if the phone is behind a NAT. If FreeSWITCH is on the same network as the phone, the phone keeps the same port. FreeSWITCH ignores that and keeps trying to contact the phone on the old port and of course fails. Only some phones seem to change their port after registration. They include 3CXPhone, X-Lite. Phones that don't do this are MicroSIP and Mizu. I have a wireshark capture file of the session from the softphone machine if anyone would like, I'll be happy to email it or publish it. So, I have three questions: 1. Why does FreeSWITCH initially send "Unauthorized" reply? 2. Does anyone know why some phones change their port during registration from behind a NAT? 3. Should I file a Jira ticket to have FreeSWITCH change UA's registered contact info when the UA sends a message with a different Contact header? Thank you *Oleg*

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Can't change external RTP IP
by Oleg Stolyar 04 Jun '13

04 Jun '13

Hi guys, I am trying to change expternal_rtp_ip for one of my sip profiles. To make thing as simple as possible, I changed it in the internal profile (internal.xml). I just set it to a literal IP address. Then I registered a soft phone on the FS and tried to make a call. The call went through and FS console said that it connected to vociemail but I could not hear anything. Wireshark said that the SDP address that FS sent back is still FS's internal IP address. Any ideas what we are doing wrong? Thank you *Oleg*

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