June 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problem with accessing HTTPs URL's via CURL with FreeSwitch.
by Venkatesh 17 Jun '12

17 Jun '12

Hi ! I have a javascript application that needs to access a web service via HTTPs. I tried using session.execute("curl", restURL) to access the same. It looks like freeswitch/mod_curl is not even making an attempt to connect to my web service when I use a HTTPs URL (verified by running tcpdump on the host when running my application). However the application works if I substitute HTTPs with plain HTTP. I wanted to know if this is a configuration issue of some sort or if this is a bug? Would appreciate any help. Venkatesh

1 0

what is the best tools for sip stress test on freeswitch?
by Samira Mh 16 Jun '12

16 Jun '12

hi, what is the best tools for sip stress test on freeswitch?

3 2

Re: [Freeswitch-users] Does mod_fifo not take in effective_caller_id_name?
by Eric Sander 16 Jun '12

16 Jun '12

Any body know if effective_caller_id_name can be passed on to modfifo. Read the wiki on it and there's no info. This would be a helpful feature to include as other parts of FS use this to display info. thanks eric_s On Thursday, June 14, 2012 at 6:35 AM, Andrew Cassidy <andrew(a)cassidywebservices.co.uk> wrote: > >mod_callcenter also ignores those variables unless you set one >called >cc_export_vars. Perhaps there's a similar mechanism for >mod_fifo? > >On 14 June 2012 05:31, wrote: > Hi All, >I have this basic dialplan: >I can see the call get logged via the first extension but it's >not >passing the info to modfifo. I can remove the 2nd extension and >put a >basic: >and the caller id name info is displayed on the phones but not when >routing through mod_fifo. I have tried both set/export and it >still >doesn't seem to work. >Is this correct? >Thanks,eric_s > >___________________________________________________________________ >______ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > Join Us At ClueCon - Aug 7-9, 2012 > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch >-users > http://www.freeswitch.org >-- >Andrew Cassidy BSc (Hons) MBCS SSCAManaging Director > > T 03300 100 960 F 03300 100 961 E >andrew(a)cassidywebservices.co.uk W >www.cassidywebservices.co.uk

1 0

A request from a user seeking a Unique Dial Plan Solution
by Sean Devoy 15 Jun '12

15 Jun '12

HI All, I have a user that wants all DID calls to go to the receptionist . BUT if she is unavailable or ON THE PHONE then go to a sequential or parallel hunt. The full request is basically ring ext 300, if unanswered in 5 seconds CONTINUE ringing 300, but add in the other extensions. At first call intercept seemed like a great choice, but if the receptionist in on a call her phone does not "ring" so nobody knows to pick it up. I was thinking <bridge> to receptionist with timeout of 5, then <bridge) to all including receptionist. However, I am concerned that she may try to answer BETWEEN to the <bridge>s and get a dial tone. Does anyone have any clever ideas? Thanks, Sean

3 3

Having trouble calling a 4-agent queue (the 4th agent never gets called)
by Yungwei Chen 15 Jun '12

15 Jun '12

Hi, I am trying to test routing calls to a queue with 4 agents using mod_callcenter. Here're the settings. * the contact of each agent is set to {leg_timeout=22}sofia/gateway/p1/12341234 (a phone number that will never be answered) * the max-wait-time of the queue is set to 88 seconds (22 * 4) * the strategy of the queue is set to sequentially-by-agent-order * each agent is assigned a unique position * the status of each agent is Available The problem is that mod_callcenter doesn't try to reach the 4th agent for some reason. If the max-wait-time of the queue is set to 98 seconds, the 4th agent will still be ignored and the first agent will be called twice (it starts over). Any ideas why? Thanks.

1 0

Best place to set variables?
by ocset 15 Jun '12

15 Jun '12

Hi In the FreeSWITCH-Cookbook, there is the following example <extension name="ring_sequentially"> <condition field="destination_number" expression="^(2001)$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data={ignore_early_media=true}sofia/internal/userA(a)local.pbx.com"/> <action application="bridge" data={ignore_early_media=true}sofia/internal/userB(a)local.pbx.com"/> </condition> </extension> Is there any reason why the "ignore_early_media=true" was not set before the bridge commands as is done with the "continue_on_fail" and "hangup_after_bridge" variables? Example: <extension name="ring_sequentially"> <condition field="destination_number" expression="^(2001)$"> <action application="set" data="continue_on_fail=true"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="ignore_early_media=true"/> <action application="bridge" data=sofia/internal/userA(a)local.pbx.com"/> <action application="bridge" data=sofia/internal/userB(a)local.pbx.com"/> </condition> </extension> Thanks O

3 4

FreeSWITCH and netvoip.ch
by sunsus 15 Jun '12

15 Jun '12

Hello I've tryed to use netvoip.ch as a gateway. But I have a lot of connection problems. Because netvoip has a load balancing between two servers sip.netvoip.ch (sip-1.netvoip.ch and sip-2.netvoip.ch). Now the problem is if you are getting with the INVITE to the first server you're getting a 304 Unauthorized with the nonce of the first server. For some reason FreeSWITCH is now sending a INVITE with the digist Authentication to the second server and this server is now answering with 304 Unauthorized with an other nonce. FreeSWITCH now ends the call with 500 internal server error. Is this a FreeSWITCH related issue or a netvoip related issue? Or is there any configuration in the gateway to say to use the same server for the authentication process? Regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-and-netvoip-ch-tp7… Sent from the freeswitch-users mailing list archive at Nabble.com.

7 18

Painted into a corner: outbound-codec-prefs, late negotiation, and inherit codec
by Jeremy Childs 15 Jun '12

15 Jun '12

A quick introduction to what I'm trying to do: I have two SIP profiles, "internal" (port 5060) and "offpremise" (port 5070). Both of these profiles are recording registrations to the same registration db, so they can be reached by user/nnn. When a device leaves the local LAN and heads out into the field, firewall port translation maps port 5060 to 5070, so FS "knows" the extension is now off-premise. All profiles on the box are set to inbound-late-negotiation and inherit-codec, to reduce transcoding. What I would like to do is set a very restrictive list of codecs on the "offpremise" profile, so that an exension registered to "offpremise" only sees a set of low-bandwidth codecs. To facilitate this, I have set outbound-codec-prefs on the profile. After much testing, I have determined that "outbound-codec-prefs" in this configuration ends up ADDING codecs to the offer, and will not remove any codecs offered by the A-leg. Interestingly, inbound-codec-prefs seems to remove codec from the A-leg offer. So, my question: is there any way of restricting the list of codecs offered on the B-leg using the profile configuration (and not from within the dialplan)? If there is any answer at all, I'd suspect it's one of the following: * a global setting that changes the behavior of outbound-codec-prefs to be restrictive, instead of additive (the source doesn't seem to suggest this) * a way to set absolute_codec_string (or other variables) from within the profile itself * a way to set a variable (again, likely absolute_code_string) to the registration instance so the codec list can be explicitly specified * any other far-out ideas for accomplishing this kind of seamless "roaming" Any ideas or hints greatly appreciated.

1 0

Problem passing and playing audio files into a conference
by Peter Stevens 15 Jun '12

15 Jun '12

Hi all, I hope that someone can help with solving the following problem. I have tried using the example code at the bottom of http://wiki.freeswitch.org/wiki/Mod_conference page, but stripped back to not record (so as to limit the number of possible problems) but just to play back selected wav files. Q: How to create a dialplan to have callers speak their name before joining the conference and announce that to other participants when they join? This is the code I'm using...having added a couple of delays <extension name="Record Name and schedule conf announce"> <condition field="destination_number" expression="^55(3\d\d\d)$"> <action application="answer"/> <action application="playback" data="ivr/ivr-call_being_transferred.wav"/> <action application="sleep" data="1000"/> <action application="log" data="INFO Caller Id Name is: ${caller_id_name}"/>    <action application="set" data="res=${sched_api (+2 none conference $1-${domain} play file_string://${sound_prefix}/conference/8000/conf-welcome.wav!${sound_p refix}/conference/8000/conf-has_joined.wav)}"/> <action application="sleep" data="1000"/> <action application="transfer" data="$1 XML default"/> </condition> </extension> Using this line, instead of the original: <action application="set" data="res=${sched_api (+2 none conference $1-${domain} play file_string://${sound_prefix}/conference/8000/conf-welcome.wav!${sound_p refix}/conference/8000/conf-has_joined.wav)}"/> with the () brackets as shown at least set the res ok, i.e. I didn't get this error message: EXECUTE sofia/nat/1101@DOMAIN set(res=) 2012-06-14 09:56:47.493255 [DEBUG] mod_dptools.c:748 sofia/nat/1101@DOMAIN SET [res]=[UNDEF] (note that I have removed the domain name and replaced it with DOMAIN in these messages) but got this instead: 2012-06-15 16:24:55.713235 [DEBUG] switch_scheduler.c:214 Added task 6 sched_api_function (none) to run at 1339773897 EXECUTE sofia/nat/1209@DOMAIN set(res=+OK Added: 6) 2012-06-15 16:24:55.713235 [DEBUG] mod_dptools.c:748 sofia/nat/1209@DOMAIN SET [res]=[+OK Added: 6] which appears to be some improvement! As before the call got transferred to the conference, but didn't run the additional audio clips specified in the action application set up line shown above and then ran the hold music. Instead I got the following invalid file format error message from file_string: 2012-06-15 16:24:57.481975 [ERR] switch_core_file.c:71 Invalid file format [file_string] for [/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-welcome.wav!/o pt/freeswitch/sounds/en/us/callie/conference/8000/conf-has_joined.wav]! 2012-06-15 16:24:57.481975 [DEBUG] mod_commands.c:2391 Command conference(3000-132.185.142.18 play file_string:///opt/freeswitch/sounds/en/us/callie/conference/8000/conf-w elcome.wav!/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-has_ joined.wav): (play) File: file_string:///opt/freeswitch/sounds/en/us/callie/conference/8000/conf-w elcome.wav!/opt/freeswitch/sounds/en/us/callie/conference/8000/conf-has_ joined.wav not found. 2012-06-15 16:24:57.481975 [DEBUG] switch_scheduler.c:138 Deleting task 6 sched_api_function (none) 2012-06-15 16:24:59.473199 [DEBUG] mod_local_stream.c:346 Opening Stream [moh/8000] 8000hz The two particular audio files conf-welcome.wav and conf-has_joined.wav do exist at the specified location. >From what I have seen on various posts in the lists, the file_stream syntax seems to be correct, so what might the problem be parsing and playing these files once into the conference? Thanks for any help Peter http://www.bbc.co.uk/ This e-mail (and any attachments) is confidential and may contain personal views which are not the views of the BBC unless specifically stated. If you have received it in error, please delete it from your system. Do not use, copy or disclose the information in any way nor act in reliance on it and notify the sender immediately. Please note that the BBC monitors e-mails sent or received. Further communication will signify your consent to this.

1 0

AMR and AMR-WB support
by Jakson Kalsson 15 Jun '12

15 Jun '12

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Freeswitch support these codecs and RFC4867 ? Also, any other Server/PBX which support AMR, AMR-WB recommended are welcome. Best regards,

2 1