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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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FS Choking on a 200 Response w/ a Duplicate "From" Header
by JP 05 Jun '13

05 Jun '13

I thought I would bounce this off the mailing list before filing a Jira ticket. I'm facing an issue where FS doesn't acknowledge a 200 OK response with a duplicate "From" header: recv 422 bytes from udp/[10.191.210.149]:5060 at 17:21:22.634270: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 10.191.210.147:5080;rport;branch=z9hG4bKNv1g56Djj6BKj From: "" <sip:0000000000@10.191.210.147>;tag=By8FpSr2m6NvF From: "" <sip:0000000000@10.191.210.147>;tag=By8FpSr2m6NvF To: <sip:1000@10.191.210.149:5060>;tag=1 Call-ID: 1efaef87-48a7-1231-d08a-000c29ccde0a CSeq: 44890363 INVITE Contact: <sip:10.191.210.149:5060;transport=UDP> Content-Type: application/sdp Content-Length: 0 Although I've not yet stepped through the code, it is my suspicion that the Sofia SIP stack is not accepting the response. Aside from the re-transmissions of the response that I'm seeing in the logs, there are no other error, info, or trace statements coming from any other FS module. RFC 3261 seems a bit vague in my opinion as to whether or not this is even allowed. As a side note, this issue came up while doing integration testing with OpenSER 1.3.4. OpenSER seems to stick in the duplicate From header whenever it does a serial proxy to multiple carriers and the first carrier leg times out. I'm wondering if anyone on the mailing list was familiar with this issue and perhaps knew of some fix/workaround on the FreeSWITCh end. Thanks, JP

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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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