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Core dump: Seg 11 if phones get powered off
by Faraz R. Khan 16 Jul '08

16 Jul '08

This seems to happen nightly. Let me explain the scenario, this is a third world country and power outages happen regularly. The UPS on which the phones are may die (and i'm sure it does). Seems like freeswitch segfaults when it cannot find these phones! (This is just my analysis- as freeswitch runs perfectly otherwise) I have a core file and have generated a backtrace from gdb. I have uploaded to JIRA as FSCORE-155. I am unclear as to whether this is a core issue or a sofia_sip issue. Please let me know if there is a workaround to this problem, as right now i have to restart freeswitch in the morning :) -- Faraz R Khan Chief Architect Emergen Consulting Pvt Ltd +92.21.529.0381 x200 www.emergen.biz

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slow hangup detection using FXO into voicemail application
by John Wehle 16 Jul '08

16 Jul '08

If I call into the voicemail application from a VoIP phone and hangup while the welcome message is playing FreeSWITCH reports the hangup right away. If I call into the voicemail application from a A204DX FXO line and hangup while the welcome message is playing FreeSWITCH doesn't report the hangup until 33 seconds have elapsed. I've emailed Sangoma however thought to post it here in case it rings a bell. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Spidermokey_curl problem? Sanity check please
by Simon Tang 16 Jul '08

16 Jul '08

Hello, I've been making Javascript curl requests and have found that larger requests sometimes come back incomplete. This is what I am doing in javascript: ------------ function callback(string, arg) { console_log("info", "returned body:" + string + "\n"); return true; } var curl = new CURL(); var blah = curl.run("GET", url, args, callback, arg, ""); -------------- comparing 'string' with tcpdump, I found that 'string' is definitely incomplete vs. tcpdump. I've looked into spidermokey_curl.c and found this code (excerpts): static size_t file_callback(void *ptr, size_t size, size_t nmemb, void *data) {...} curl_easy_setopt(co->curl_handle, CURLOPT_WRITEFUNCTION, file_callback); curl_easy_setopt(co->curl_handle, CURLOPT_WRITEDATA, (void *) co); I had the suspicion that the callback is getting a partial response and passing it into the JS object as a completed response: - From http://curl.netmirror.org/libcurl/c/curl_easy_setopt.html: "The callback function will be passed as much data as possible in all invokes, but you cannot possibly make any assumptions. It may be one byte, it may be thousands. The maximum amount of data that can be passed to the write callback is defined in the curl.h header file: CURL_MAX_WRITE_SIZE." >From this, I think that it is the callback's responsibility to ensure that the complete response was received and to save the data somewhere. To confirm my suspicion, I replaced the callback function in spidermonkey_curl.c with vanilla logging and took out the JSAPI references: static size_t file_callback(void *ptr, size_t size, size_t nmemb, void *data) { register unsigned int realsize = (unsigned int) (size * nmemb); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,"BODY STUFF::::::%s\n", (char*) ptr); return realsize; } I recompiled and reproduced the offending http request. I found that the log is called twice, with each log comprising both pieces of the complete response. When I reverted back to the original file_callback function and added a log in there, the log only recorded the first part of the response (incomplete). I'm not sure if I'm using this correctly or if my analysis is correct, so if someone can provide me some feedback, that'd be great. If it's a legit bug, I can file it. Simon Tang

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send_dtmf problems
by John Wehle 16 Jul '08

16 Jul '08

> <action application="gentones" data="1234567890"/> Yep, this works through the A204d FXO openzap lines, though sometimes there's little odd click / hiccup in the middle of playing the tones. I'm a little confused as to why using send_dtmf didn't seem to work well, however no matter. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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HOW TO PLAY MUSIC DURING THE CALLS ?
by M.Emran 15 Jul '08

15 Jul '08

Please help me to configure FS to play music during the calls. -- Regards ---------- M Emran Managing Director InSpiration Software Ltd.

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ClueCon in 3 WEEKS
by Anthony Minessale 15 Jul '08

15 Jul '08

I wanted to let everyone know that ClueCon 2008 is just under 3 weeks away! Everyone who is planning to attend: Please make sure you are registered and have your rooms booked. We have an overflowing schedule of speakers but we could still probably squeeze in a few more lightening talks if anyone wants to speak. This year is shaping up to be the best yet with developers or reps from most of the popular Open Source VoIP applications represented including: FreeSWITCH(tm) (naturally) ;) OpenSER OPAL sipX/SIPfoundry Asterisk Yate Bayonne SofaSWITCH We also have reps or developers from several hardware vendors including: Sangoma PIKA Digium Dialogic SNOM OCTASIC We also have 2 professional voice firms attending including: GM Voices (creator of the famous Cepstral Callie and all the FreeSWITCH(tm) prompts) Allison Smith (The voice of Asterisk(tm)) -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Playing an audio file to end destination before bridging the call
by Adnan Barakat 15 Jul '08

15 Jul '08

Hi All, I'm looking for a way to play an audio file to the end destination before bridging the call, eg. below is an example in Asterisk which I'm trying to replicate in FS. exten => _0ZXXXXXXXXX,1,Dial(SIP/outsip/${dest},30,rA(/srv/files/whisperfile)) Is this possible in FS? Any feedback would be highly appreciated. Adnan

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send_dtmf problems
by John Wehle 15 Jul '08

15 Jul '08

> Try a sleep after the answer. Will do ... just curious as to why and for how long? >> 1) When I dial the extension from a Grandstream GXP-2000 ... >> How do I configure FreeSWITCH to send both RTP digits and inband audio? > The VoIP phone has no reason to reproduce any DTMF. I understand that the VoIP phone has no reason to create DTMF tones from the RTP digits sent by FreeSWITCH. What I was asking is how to you configure FreeSWITCH to generate inband DTMF tones in addition to the RTP digits. I.e. this particular phone can be configured to send DTMF digits to FreeSWITCH as any combination of inband audio, RTP digits, and / or SIP info. Going in the opposite direction I see where in the FreeSWITCH sofia xml configuration type you can set the dtmf-type as either rfc2833 or as SIP info however I don't see an option for using both nor do I see an option for generating inband DTMF audio. Note: I was just using send_dtmf with the VoIP phone for testing purposes ... this application doesn't actually need to send DTMF to a VoIP phone. >> 2) When I dial the extension from a FXO port on a Sangoma A204D it connects, > Is this using OpenZAP? Yes. In my particular application the System 25 PBX connects to FreeSWITCH using OpenZAP running on a Sangoma A204DX card. At the end of a call sent to voicemail I need FreeSWITCH to send some DTMF tones to the PBX in order to turn on / off the message waiting indicator. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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outbound fxo line pooling
by John Wehle 15 Jul '08

15 Jul '08

> a or A depending on if you want top down.. or bottom up. Okay, so with OpenZAP I can use either a number to refer to a specific port or A to refer to any available port in a span. Other than using a number or A (and a) OpenZAP currently has no mechanism for defining explicit pools / port ranges. So to do something like have a four port card setup with inbound calls using all four ports and outbound calls using only ports 2 - 4 would I need to configure the card as two spans with span 1 containing port 1, and span 2 containing ports 2 - 4? Then I route outbound calls to span 2 using the A port (aka any available port on span 2)? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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SIPfoundry cooperates with the FreeSWITCH project
by Brian West 15 Jul '08

15 Jul '08

Check it out!! http://digg.com/software/SIPfoundry_cooperates_with_the_FreeSWITCH_project Check it out... Digg it and pass it around to 10 friends. ;) /b Brian West sip:brian@freeswitch.org

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