July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Cisco CUCM 6.1 and Freeswitch SIP trunk
by David Martin 26 Jul '11

26 Jul '11

All, Firstly apologies if this is not the correct place to ask, but I am having problems getting CUCM 6.1 to talk with freeswitch over a SIP trunk (I only need inbound from Cisco -> Freeswitch). I have enabled 3pcc, but am getting the following sofia trace: http://pastebin.com/g9VmvxFe Can anyone point me in the right direction? I have also tried enabling MTP on the Cisco side, but this does not make any difference. Thanks in advance. David

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(no subject)
by Sanja Angelova 26 Jul '11

26 Jul '11

Dear all! I am new to FreeSwitch and I am trying to map a number for example 654321 to sip adress like this 1005(a)10.1.1.1 with configuring a local DNS server. When I write enum 654321 in fs_cli it gives me: Offered Routes: Order Pref Service Route ============================================================================== 10 100 E2U+sip sofia/${use_profile}/1005(a)10.1.1.1 and seems to be fine. But when I make a call from my soft phone, the FreeSwitch ends up my call and it gives me error 503 service unavailable. I have no idea what the problem is, can you please help me? Thank you in advance

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sipx conference API
by Jayakrishnanu 26 Jul '11

26 Jul '11

Hi Folks Is there any way to add a person in to existing conference? > I configured sipx and I successfully made calls through call controller > API > but when I tried to implement conference web service API I could get the > response from the argument xml_list,volume_in,kick,lock but when I tried > to dial a number in to the conference using "curl --digest -k -X PUT > https://userid:Password@hostname:8085/conference/conference- > name/dial/XXXXX/dial-number".I couldnt get what is the > <endpoint_module_name> for the same. Can anybody give me the correct url for dial service in conference and how can I figure what is the endpoint-name in my system? -- View this message in context: http://old.nabble.com/sipx-conference-API-tp32129940p32129940.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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rxfax on b-leg
by slash＠nat-tele.com 26 Jul '11

26 Jul '11

Hello! I wonder if there is a way to make rxfax recieve fax on outgoing call? The idea is that registered FS user dials out, than tone_detect is used to detect fax tone and start rxfax. The difference is that rxfax should be run on B-leg, instead of A-leg in incoming fax call scenario. I've tried to make a test config with outgoing call, rxfax & tone_detect, but when I dial out and remote user starts sending fax to me, FS detects a tone correctly, starts rxfax, but on my leg (A-leg, i hear fax tones) and B-leg (with real fax machine) is hanged up. Are there are solutions to explain somehow to rxfax that it should be run on B-leg, not on A-leg? Thanks in advance! With best regards, Kirill.

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SDP Append
by Ricardo Martinez 25 Jul '11

25 Jul '11

Hello. I’m trying to append two distinct SDP fields in the outgoing call leg from Freeswitch. Right now I’m only capable to attach only one : <extension name="add-annexb=no" continue="true"> <condition field="${switch_r_sdp}" expression="/(.*)(a=fmtp:18 annexb=no)(.*)/s"> <action application="export" data="sip_append_audio_sdp=a=fmtp:18 annexb=no"/> </condition> </extension> Because when I’m try to attach a second one, it rewrites the first one. Thanks. Ricardo Martinez.-

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Re: [Freeswitch-users] Can't get variable set in dial plan
by Joshua Foshee 25 Jul '11

25 Jul '11

That fixed it. The fix was to change hours to ${hours} in the condition field. Thanks, Josh From: Steven Ayre [mailto:steveayre@gmail.com] Sent: Monday, July 25, 2011 3:40 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Can't get variable set in dial plan Variables are not fields. Fields are a specific set of attributes of the call... you can however check variables in a condition, it just has a slightly different syntax. Try this. <extension name="public_did"> <condition field="destination_number" expression="^(.*)$"> <action inline="true" application="set" data="hours=1"/> </condition> <condition field="${hours}" expression="^1$"> <action application="answer"/> <action application="ivr" data="GAC-BH"/> <action application="hangup"/> <anti-action application="answer"/> <anti-action application="ivr" data="GAC-BH"/> <anti-action application="hangup"/> </condition> </extension> Also note I changed your regex of "1" to "^1$". The first matches any string containing a digit 1 including 10, 21, 100, 200351. The latter matches the start (^) and end ($) of the string so will only match the exact string of 1. -Steve On 25 July 2011 20:54, Joshua Foshee <Joshua.Foshee(a)logixcom.com> wrote: I am trying to set a variable in the dial plan and use it on a condition. I have pushed the variable setting as close to the condition for testing and this is what I get. Why does it not see the variable set? <extension name="public_did"> <condition field="destination_number" expression="^(.*)$"> <action inline="true" application="set" data="hours=1"/> </condition> <condition field="hours" expression="1"> <action application="answer"/> <action application="ivr" data="GAC-BH"/> <action application="hangup"/> <anti-action application="answer"/> <anti-action application="ivr" data="GAC-BH"/> <anti-action application="hangup"/> </condition> </extension> Here is the console output Dialplan: FreeTDM/1:7/405790xxx8 Regex (PASS) [public_did] destination_number(405790xxx8) =~ /^(.*)$/ break=on-false Dialplan: FreeTDM/1:7/405790xxx8 Action set(hours=1) INLINE EXECUTE FreeTDM/1:7/405790xxx8 set(hours=1) 2011-07-25 12:05:25.191357 [DEBUG] mod_dptools.c:1063 FreeTDM/1:7/405790xxx8 SET [hours]=[1] Dialplan: FreeTDM/1:7/405790xxx8 Regex (FAIL) [public_did] hours() =~ /1/ break=on-false Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action answer() Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action ivr(GAC-BH) Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action hangup() _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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

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Getting Started with Speech Recognition
by Bryan Smart 25 Jul '11

25 Jul '11

I've been attempting to experiment with this part of Freeswitch, but it seems that this is an area that hasn't yet received a lot of attention with regard to docs and samples. Perhaps a some of you that are experienced in this area wouldn't mind clearing up some of my confusion? First, I have FS working fine (outside of ASR). Pocketsphinx is built and loaded. As for me, I have some experience with VXML, and generally know how a grammar works, but the last phone ASR projects that I was involved with were several years back on Tellme Studio. First, as to grammars, I'm not sure, but it seems as though FS has used different formats at different times. The mod_pocketsphinx page says that JSGF is the current format. Can the JSGF grammar files be loaded without being compiled to another format? I ask that, as some of the examples, such as the LUA Directory example, talk about compiling grammars, and the format shown there doesn't appear exactly the same as what I see from the JSGF examples. Is the info about compiling grammars obsolete? Next, I'm not clear about the use of the detect_speech command. The wiki page lists some example forms of the command, but doesn't explain the purpose of the different forms, nor their arguments. Here are some examples: detect_speech <mod_name> <gram_name> <gram_path> [<addr>] >From the examples, I see that "pocket sphinx" can be used for the module. I think that gram_path is the path to the grammar file, without the ".gram" suffix, and uses the grammar folder as a base. However, what is addr? An address? To, or for, what? detect_speech grammaron <gram_name> I think that detect_speech grammaron <gram_name> Might be used when a large .gram file has been loaded, and the code needs to enable a grammar for a specific context. However, I don't see a command for just loading a .gram file without also specifying a grammar name. If I use this grammaron command, will it search all files in the grammar folder for grammars that match the name, or is this just for enabling a grammar that I've already loaded through other means? Maybe grammaron is used to activate a grammar that was disabled with detect_speech grammaroff <gram_name> But, if so, then what is the purpose of detect_speech nogrammar <gram_name> Does that also disable a grammar? Maybe it deletes the grammar from memory, also? For: detect_speech param <name> <value> What are the possible parameter names and acceptable values? detect_speech start_input_timers What are the purposes of the start_input_timers? Does this cause ASR to time out if no input is recognized? Does it cause ASR to pause for a time before beginning to recognize speech? I've tried to write up a simple grammar with only a few terms, and have a LUA IVR script attempt to detect speech using this grammar. My input callback is never called with any sort of speech event when I speak, though, and I don't receive any FS console errors. I must be making some incorrect assumptions about how the commands are used, or how the grammar is formatted. Perhaps my tests aren't working because there is a problem with Pocketsphinx. I can't try the Pizza demo to verify that Pocketsphinx is working correctly, as I'm unable to download its grammar while bkw.org is offline, and no other sources for the grammar come up on a Google search. Finally, I'd like to know if there are any compelling reasons to use one scripting language over another for speech-driven IVRs. I have experience with Lua, Javascript, and PHP, so could use any of them. The Freeswitch book uses Lua a lot, so that's what I've been trying so far. Perhaps I'd be better served with Javascript? Thanks for any pointers! Bryan

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Asterisk dial status
by Sam 25 Jul '11

25 Jul '11

Hi, I am trying to port over an Asterisk AGI script that does a call bridge to Freeswitch using perl, I wanted to know what is the equivalent of the asterisk dial status like: ANSWER, NO_ANSWER, CANCEL, BUSY, CONGESTION, CHANUNAVAIL ? I wanted to know if what methods can be used to gets these statues?

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Can't get variable set in dial plan
by Joshua Foshee 25 Jul '11

25 Jul '11

I am trying to set a variable in the dial plan and use it on a condition. I have pushed the variable setting as close to the condition for testing and this is what I get. Why does it not see the variable set? <extension name="public_did"> <condition field="destination_number" expression="^(.*)$"> <action inline="true" application="set" data="hours=1"/> </condition> <condition field="hours" expression="1"> <action application="answer"/> <action application="ivr" data="GAC-BH"/> <action application="hangup"/> <anti-action application="answer"/> <anti-action application="ivr" data="GAC-BH"/> <anti-action application="hangup"/> </condition> </extension> Here is the console output Dialplan: FreeTDM/1:7/405790xxx8 Regex (PASS) [public_did] destination_number(405790xxx8) =~ /^(.*)$/ break=on-false Dialplan: FreeTDM/1:7/405790xxx8 Action set(hours=1) INLINE EXECUTE FreeTDM/1:7/405790xxx8 set(hours=1) 2011-07-25 12:05:25.191357 [DEBUG] mod_dptools.c:1063 FreeTDM/1:7/405790xxx8 SET [hours]=[1] Dialplan: FreeTDM/1:7/405790xxx8 Regex (FAIL) [public_did] hours() =~ /1/ break=on-false Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action answer() Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action ivr(GAC-BH) Dialplan: FreeTDM/1:7/405790xxx8 ANTI-Action hangup()

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Callcenter MOD question
by Joshua Foshee 25 Jul '11

25 Jul '11

I am setting up a queue and I have ring all set with everything else sent to default pretty much. What I am trying to accomplish is that agents have phone that show 4 lines for their extension. If they are on the phone with an outbound call I want to skip trying to send them a call. How is this accomplished where the callcenter mod knows to not send a call to them. All agents are static assigned in the callcenter config file. Copy of config file below. <configuration name="callcenter.conf" description="CallCenter"> <settings> </settings> <queues> <queue name="sales@default"> <param name="strategy" value="ring-all"/> <param name="moh-sound" value="$${hold_music}"/> <param name="time-base-score" value="queue"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="14400"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="120"/> </queue> <queue name="collection@default"> <param name="strategy" value="ring-all"/> <param name="moh-sound" value="$${hold_music}"/> <param name="time-base-score" value="system"/> <param name="tier-rules-apply" value="false"/> <param name="tier-rule-wait-second" value="300"/> <param name="tier-rule-wait-multiply-level" value="true"/> <param name="tier-rule-no-agent-no-wait" value="false"/> <param name="discard-abandoned-after" value="60"/> <param name="abandoned-resume-allowed" value="false"/> <param name="max-wait-time" value="0"/> <param name="max-wait-time-with-no-agent" value="120"/> </queue> </queues> <agents> <agent name="513@default" type="callback" contact="[call_timeout=10]user/513(a)192.168.1.3" status="Available" max-no-answer="3" wrap-up time="10" reject-delay-time="10" busy-delay-time="60" /> </agents> <tiers> <tier agent="513@default" queue="sales@default" level="1" position="1"/> <tier agent="513@default" queue="collection@default" level="1" position="1"/> </tiers> </configuration>

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