November 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

SVN r10539
by Michael S Collins 26 Nov '08

26 Nov '08

Brian, So all you did was create a dp entry named "set_openzap_domain" and have it match on two separate conditions: ${domain_name} is empty and source is "mod_openzap". The only thing it does is to set the domain name to the default domain. And, since you don't want this entry to be the only thing that happens on openzap calls, you have continue set to true. Do I grok it or do I need to get back on the short yellow bus? :) -MC Sent from my iPhone

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Pocketsphinx logs of recognizer and recordings kept were in FS? CMN setting & -cmninit option help.
by mszlazak＠aol.com 26 Nov '08

26 Nov '08

Were are the PocketSphinx's recognizer and recordings logs kept in Windows version of FS? Has anyone tried to getting the proper cmn value from looking at these logs and added -cmninit option to the recognizer? Mark. ?

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switch_ivr_set_user() can't find user
by John Wehle 26 Nov '08

26 Nov '08

> This is the key to why its not working source is mod_openzap so might Yes ... I understand that. I'm pretty much running the supplied sample configuration which apparently doesn't handle openzap. Is there a specific JIRA category I should use to log this issue? I took a quick glance and wasn't sure what to use. -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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switch_ivr_set_user() can't find user
by John Wehle 26 Nov '08

26 Nov '08

> No domain != domain_name they are different. Well ... yes and no. default.xml has: <extension name="set_domain" continue="true"> <condition field="${domain_name}" expression="^$"/> <condition field="source" expression="mod_sofia"/> <condition field="${sip_auth_realm}" expression="^$"> <action application="set" data="domain_name=$${domain}"/> <anti-action application="set" data="domain_name=${sip_auth_realm}"/> </condition> </extension> so it seems that it's intended for domain_name to equal domain if a realm isn't supplied. If I use: <extension name="set_domain" continue="true"> <condition field="${domain_name}" expression="^$"/> <condition field="source" expression="mod_sofia"> <action application="set" data="domain_name=${sip_auth_realm}"/> <anti-action application="set" data="domain_name=$${domain}"/> </condition> </extension> then things work. Thoughts: 1) It may be desirable to be able to specify the domain associated with an openzap line in the openzap.conf.xml file. 2) The comment by set_domain says:  However it appears that the logic is wrong. It fails to handle cases where the source isn't mod_sofia. What JIRA category should I file this under? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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switch_ivr_set_user() can't find user
by John Wehle 26 Nov '08

26 Nov '08

> make sure you have the domain_name variable set so the call can go to > the right domain name. Set where? I'm pretty much using the stock sample files. vars.xml contains: <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> It was my understanding that domain defaults to the IP address of the server's interface. openzap.conf.xml contains: <param name="context" value="default"/> in each of the analog_spans / analog_em_spans. Is something else needed to specify the domain for processing inbound openzap calls? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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switch_ivr_set_user() can't find user
by John Wehle 26 Nov '08

26 Nov '08

I can call between VoIP phone ext 1001 and 1003 fine. I can call from VoIP phone ext 1003 over a winkstart line into the PBX fine. Before updating to the current SVN I could also call from the PBX over a winkstart line to VoIP ext 1003 fine. Now what I see is: [NOTICE] switch_channel.c:551 switch_channel_set_name() New Channel OpenZAP/4:1/1003 [d99f59f2-52bb-dd11-80a4-001fc6ab49e2] [WARNING] switch_ivr.c:1832 switch_ivr_set_user() can't find user [default@] [WARNING] mod_dptools.c:2024 user_outgoing_channel() Can't find user [1003@] [ERR] switch_ivr_originate.c:1067 switch_ivr_originate() Can not create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] It appears openzap sees the request from the PBX fine ... somehow FreeSWITCH can't connect the openzap inbound call to 1003 with the VoIP phone on ext 1003. Suggestions / pointers? -- John ------------------------------------------------------------------------- | Feith Systems | Voice: 1-215-646-8000 | Email: john(a)feith.com | | John Wehle | Fax: 1-215-540-5495 | | -------------------------------------------------------------------------

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Re: [Freeswitch-users] Javascript: record ringing of session
by Michael S Collins 25 Nov '08

25 Nov '08

Sent from my iPhone On Nov 25, 2008, at 6:03 AM, Birgit Arkesteijn <birgit(a)westhawk.co.uk> wrote: > Hi Michael, > > Sorry to nag you again. > I'm getting close to my deadline and I would like to give our customer > some feedback on this issue. > > Would you be so kind to explain what the issue is with recording rings > when doing 'new Session(originate_url)'? I don't know for sure but I suspect this doesn't return while there is early media, therefore you don't actually get the ringing. > > > As far as I've been able to work out, this statement will only > return after > a) the (or some) timeout has expired or > b) the user has picked up > > Only in case b) we've got a originated session, that can be used to > record. However, since the user has already answered, the rings are > not > recorded. > > Is that right? That is my understanding, yes. > > > Cheers, Birgit > Can you experiment with this in a regular dialplan outside of js? Try creating a simple dp entry that does these apps: pre_answer record_session info sleep (20 seconds or so) hangup Say you create the above dp entry as extension 2500. Do an originate at the CLI and examine the results: originate sofia/gateway/westhawk/0663 2500 This should call the URL and as soon as there is early media (hopefully ringing) it will put the originated call leg into ext 2500 which will pre answer the call (so you can "hear" the early media), execute the record session, do an info dump for good measure, sleep for a while and then hang up. You should then be able to listen to the sound file and if all goes well you will hear ringing prior to the far end answering. The results of this test will help you decide where to go next. :) -MC > F > On 24/11/08 11:11, Birgit Arkesteijn wrote: >> Hi Michael, >> >> Thanks for your effort and feedback. >> >> I dial multiple endpoints, but I dial them sequentially, i.e. I will >> only dial the next merchant, when the previous one hangs up. >> In this scenario, each "task" always only involves two channels: one >> consumer and one merchant. >> >> One potential complication is that after the merchant picks up, I'll >> transfer (bridge?) him/her to a conference call where the consumer is >> waiting. I would like the recording to continue after this bridging. >> >> "Long live FreeSWITCH!" ;-) >> >> Cheers, Birgit >> >> >> On 21/11/08 01:05, Michael Collins wrote: >>> Birgit, >>> >>> After reviewing my dialplan I realize now that what I'm doing won't >>> work for the scenario you've got, specifically because I am never >>> dialing multiple endpoints. I'm going to ruminate on this for a >>> little >>> while and see what I can think of. I'm sure there's a way to do it. >>> "If FreeSWITCH can't do it, nothing can!" :) >>> >>> -MC > > > -- > -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, > -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK > -- Company no: 1769350 > -- Registered Office: > -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. > -- tel.: +44 (0)161 237 0660 > -- <URL: http://www.westhawk.co.uk> > > _______________________________________________ > 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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Troubles with FreeSWITCH and Sangoma A104
by Evgeniy Zolotov 25 Nov '08

25 Nov '08

1. We have successfully configured A-104 + WANPIPE-3.3.14 + FreeSWITCH( over mod_openzap) under CentOS 5.2. But we cann't unite SVwanpipe-i386-5.10.pkg + SVzaptel-i386-5.10.pkg + FreeSWITCH( over mod_openzap ) under Solaris 5.10 Does it possible????? Do we need any other packages for SunOS? 2. This problem is expressed that FreeSWITCH's function tdmv_api_open_span(), which makes open() for path /dev/wptdms%dc%d, where %d - is a span & chun, returns "failure configuring device". Under CentOS 5.2 it works perfect, but under Solaris 5.10 in doesn't work. Does it means, that FreeSWITCH or Sangoma's equipment cann't work under Solaris 5.10?

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Javascript: record ringing of session
by Birgit Arkesteijn 25 Nov '08

25 Nov '08

Hi, In Javascript, I do the following: var mSession = new Session("{ignore_early_media=true,originate_timeout=8}" + endpoint_url); mSession.execute("record_session", recordfile); This works very well, however, the client would like the rings to be recorded as well. I've tried using various channel variables, but 'new Session' only returns after the call is answered. I tried calling apiExecute("record", <file_path>); before 'new Session', but that returns an error. Is is possible to record the rings (in javascript)? And if so, how would I do that? Thanks, Birgit -- -- Birgit Arkesteijn, birgit(a)westhawk.co.uk, -- Westhawk Ltd, Albion Wharf, 19 Albion Street, Manchester M1 5LN, UK -- Company no: 1769350 -- Registered Office: -- 15 London Road, Stockton Heath, Warrington WA4 6SJ. UK. -- tel.: +44 (0)161 237 0660 -- <URL: http://www.westhawk.co.uk>

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Big problems with disconnects
by Dennis 25 Nov '08

25 Nov '08

hi, i have big problems with disconnects, when bridging and unbridging calls. because i had random diconnects when testing fs from one softphone to the other, i set up a little dialplan (socket outbound), to do some hardcore testing. after the inbound is answered, i do an originate to an outbound. when the outbound answers, i bridge the inbound and outbound. then i do uuid_transfer, 5 seconds later a uuid_bride, 5 seconds later a uuid_transfer and so on and so on. latest after 5 minutes of switching, the connection seems to hang for about 29 seconds and then ends. this ALWAYS happens after the debug message: switch_core_state_machine.c:187 switch_core_standard_on_park() sofia/internal/98765 Standard PARK see http://pastebin.freeswitch.org/6281 at line 82. i repeated that for at least 30 times - always with the same result. what could be the problem? thanks

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