September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

gateway setup check
by Nigel Kent 24 Sep '10

24 Sep '10

I want to check my gateway setup and that calls are able to be sent originate sofia/gateway/xxx.xxx.xxx.xxx/0044207343 100 -ERR INVALID_NUMBER_FORMAT xxx.xxx.xxx.xxx is a public ip address and the gateway name is called the same <gateway name="xxx.xxx.xxx.xxx"> Why do I get invalid gateway error? 2010-09-23 20:43:34.756883 [ERR] mod_sofia.c:3603 Invalid Gateway freeswitch@internal> 2010-09-23 20:43:34.756883 [NOTICE] mod_sofia.c:3899 Close Channel N/A [CS_NEW] 2010-09-23 20:43:34.756883 [DEBUG] switch_core_state_machine.c:428 () Running State Change CS_DESTROY 2010-09-23 20:43:34.756883 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY 2010-09-23 20:43:34.756883 [DEBUG] mod_sofia.c:350 N/A SOFIA DESTROY 2010-09-23 20:43:34.756883 [DEBUG] switch_core_state_machine.c:438 (N/A) State DESTROY going to sleep 2010-09-23 20:43:34.756883 [ERR] switch_ivr_originate.c:2491 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2010-09-23 20:43:34.756883 [DEBUG] switch_ivr_originate.c:3310 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]

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mod_xml_cdr with offline web server
by Max Clark 24 Sep '10

24 Sep '10

Hello, I've been looking at using mod_xml_cdr posting to a web form. Question: what happens to freeswitch if the web server goes offline? Will calls still process? Thanks in advance, Max

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group_confirm_cancel_timeout on multiple legs
by Neil Burgess 24 Sep '10

24 Sep '10

Hi there, I have a dial string similar to below, which dials two SIP URIs, one after another using leg_delay_start of 0 and 5, and leg_timeout of 10 and 10 to stagger the calling pattern. Both SIP URI's have a group_confirm_file associated with them, where pressing a digit will cause the call to be connected! {group_confirm_key=exec,group_confirm_file='javascript GroupConfirm.js',ignore_early_media=true, ,group_confirm_cancel_timeout=true}[ leg_delay_start=0,leg_timeout=10,rmvLegConfirm=true]sofia/internal/12005@blah.com;fs_path=sip:172.30.98.31,[leg_delay_start=5,leg_timeout=10,rmvLegConfirm=true]sofia/internal/12003blah.com;fs_path=sip:172.30.98.31<mailto:sofia/internal/12005@blah.com;fs_path=sip:172.30.98.31,%5bleg_delay_start=5,leg_timeout=10,rmvLegConfirm=true%5dsofia/internal/12003blah.com;fs_path=sip:172.30.98.31>) I have set group_confirm_cancel_timeout=true to ensure that the call doesn't drop due to timeout when answered and in the confirm stage. I have two scenarios relating to timeout problems; both of which I'm sure are the same! Scenario 1 When a call is made, we wait for 7 seconds until both extensions ring, then pick up the call on first extension. FS plays the group confirm message and doesn't timeout as expected. However the second phone then continues to ring beyond its configured leg_timeout? Question: does group_confirm_cancel_timeout cancel all leg timeouts as soon as one extension answers? If so, is there a way to make it only apply to the leg that answered, and allow other legs to carry on with their existing timeouts. Scenario 2 When a call is made, we allow extension 1 to start ringing and pick it up. It plays the group confirm message and doesn't timeout as expected. The second extension never starts ringing however, whilst the group_confirm phase is active. This means that if the first extension is rejected in the group_confirm phase, the second extension never gets going! Again, is there a way to make the group_confirm_cancel_timeout apply to the single leg and not all legs in the originate! Any ideas please... Neil

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freeswitch and git problem/question
by covici＠ccs.covici.com 24 Sep '10

24 Sep '10

Hi. I ran into a strange problem with git yesterday -- I had checked out an older commit using git co and the commit number for testing, but git would not let me go back to head -- it complained that I was not in a branch and wanted me to specify the repository and refs on the command line, but nothing I did worked, so I wiped the whole thing out and started again. However there has got to be a way to do this -- I am sure I did things like that in other git projects, so what am I doing wrong because I don't know git as well as I would like to. Thanks for any ideas on this. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Problem with transfer/refer/hold
by Helmut Kuper 24 Sep '10

24 Sep '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, today I found a problem with Snom transfer function. What I do is this: 1. Snom A calls Snom B 2. Snom B icks up and putting Snom A on hold 3. Snom A hears hold music 4. Snom B calls Snom C 5. Snom C picks up 6. Pressing Transfer on Snom B 7. Pressing Transfer on Snom B again :) 8. Connection to Snom B is terminated by FS 9. Snom A hears nothing 10. Snom C hears hold music I found that Snom B sends an HOLD INVITE to Snom C when transfer is pressed for the 2nd time on Snom B. I opened a Bug at Snom, but I'm not sure whether this is also a FS issue ... If you need some pcap traces, I can send you all I have about this. I use Snom FW 8.4.18 and .20 and FS GIT head of 2010-08-26 (Unfortunately that revision has a .version file in which only contains "git-") regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFMmHiD4tZeNddg3dwRAskVAKC3OnIBoPaIIW7yhAs/VuAi7X14sgCgn4Uw VpfSKkOGMRcHh06ishvAvl4= =xrkk -----END PGP SIGNATURE-----

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[INFO] mod_dialplan_xml.c:331 Processing Unavailable <Restricted>->2901 in context default
by Mario 24 Sep '10

24 Sep '10

I ran some traces and can't seem to figure out if this is an error I can fix. Otherwise it works. No luck on web searches so far. Thanks. [INFO] mod_dialplan_xml.c:331 Processing Unavailable <Restricted>->2901 in context default My 2901 dial plan is: <extension name="uid1vm"> <condition field="destination_number" expression="^2901$"> <action application="answer"/>  <action application="voicemail" data="default ${domain_name} 200"/> </condition> </extension>

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Re: [Freeswitch-users] limits on text to speech output length
by Daniel Neubert 24 Sep '10

24 Sep '10

It's not only good for having a better scalable infrastructure - it also gives you the ability to freely change your tts engine or add functionality like speech recogniction or recording. Daniel Am 24.09.2010 05:59, schrieb A. Lester Buck III: > Dear Daniel, > > Thanks so much for the pointer. I'll certainly check that out. I don't > really have much of a scaling issue at the moment, but I sure hope I do > someday! Always good to keep the architecture as small parts loosely > joined. > > Best regards, > > Lester > > >> Hi, >> >> instead of calling swift directly you can consider calling it via an >> MRCP server like UniMRCP. >> >> It's all free software and works like a charm in my setup. I'm using >> mod_unimrcp on our FreeSWITCH servers and am running UniMRCP as a >> server on a single node that is our dedicated speech server. >> >> That scenario also offloads the task of speech generation to another >> machine to save resources on the FreeSWITCH nodes. If you use other >> cepstral languages than english you need to manually patch the sources. >> >> Daniel >> >> Am 23.09.2010 22:57, schrieb A. Lester Buck III: >>> Hi, >>> >>> I'm running Freeswitch built from trunk a week or so ago, 1.0.head. >>> >>> I am using a Cepstral voice for text to speech. I generate a long >>> string of text, including SSML breaks, that is rendered to speech by the >>> swift utility packaged with Cepstral voices. The test string I built >>> renders as 1m:46s of speech, and a .wav file about 1.6MB, from the swift >>> command line. >>> >>> I changed single quotes to double quotes and escaped them, as mentioned >>> on the wiki page for mod_cepstral, and I now have a working text to >>> speech path using ESL built for Ruby 1.9.2, using the sample Ruby code >>> on the wiki. >>> >>> Everything basically works fine, even on my test machine which is a >>> $5/mo Xen VPS with 10ms kernel timer resolution. (lowendbox.com) >>> >>> My issue is that the speech cuts off midway through the call, when I >>> guess some limit on generated output is being reached. Before I dig >>> into the guts of mod_cepstral and Freeswitch, could someone point me to >>> what might be limiting the length of output? Swift seems to generate a >>> wav file of indefinite length, so at least the issue is somewhere in the >>> open source parts of the server. >>> >>> Having yet looked at the guts of mod_cepstral, I presume that a >>> high-performance output module does not have the swift utility write a >>> .wav file to disk and then play it back, but instead lets swift shove >>> the bytes directly into a socket. What is a good Freeswitch example of >>> this architecture? >>> >>> >>> Thanks, >>> >>> Lester >>> >>> _______________________________________________ >>> 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 >> ATT00001.txt >> >> >> _______________________________________________ >> 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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CODEC2 Support in FreeSWITCH
by Brian West 23 Sep '10

23 Sep '10

I have been working on adding CODEC2 support to FreeSWITCH which is a low bitrate LGPL codec by David Rowe. I have made changes to create the lib required to make this work and waiting on them to be pushed back up stream so I have checked the lib into the libs folder for now. You can find the lib and module in the latest git pull. ./bootstrap.sh ./configure make mod_codec2-install edit modules.conf.xml to load the mod add the codec to your allowed codec list on sofia profiles. Please check it out and let me know what you think. Thanks, Brian

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Registration of SIP client not lasting
by bsd 23 Sep '10

23 Sep '10

Hello, I am using FreeSwitch embedded in a pfSense device. Things are running ok… My SIP client is Acrobits SoftPhone running on IPhone 4g. The Sip client is registering ok, but when I look at the registration on FreeSwitch, it disappears after a couple of minutes… Is there a way for me to tune that somewhere at the level of FreeSwitch ? Thanks. ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯ Gregober ---> PGP ID --> 0x1BA3C2FD bsd @at@ todoo.biz ¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯¯

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Voicemail not to announce extension number
by mazilo 23 Sep '10

23 Sep '10

I have used the examples/conf files to configure my FS (v1.0.6) hosted on a Seagate DockStar running on an OpenWRT OS. So far, so good. I am able to place/receive outbound/receive calls, respectively sans any problems. The Voicemail (VM) nicely kicks in when no one answer the phone and annouces the person at extension XXXX is not available ... It certainly will be nice if anyone here can help me to modify the VM not to announce the extension number. ----- don't and stop are the ONLY two 4-letter words considered offensive to men, but not when used together. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Voicemail-not-to-announce-ext… Sent from the freeswitch-users mailing list archive at Nabble.com.

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