April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Picking up voicemail
by Dave Stevenson 01 May '10

01 May '10

Hi, I have FreeSWITCH running over an internal IP network, connected to the PSTN line with an SPA-3102. When an outside (PSTN) call comes in, a Group of phones are rung until the first one picks up or until the caller falls over to voice mail and can leave a message in the Group Mailbox (100). I can retrieve the messages from any local phone - so far, so good. I'd like to be able to call in through the same PSTN line and interrogate the group mailbox, or any mailbox come to that. Can someone point me in the right direction for how to set this up please ? i.e., once FreeSwitch has answered the call and the voicemail prompts have stated, can they be interrupted so that I can get to the voice mail extension. regards Dave

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Lag on extension behind spa3102
by guru singh 01 May '10

01 May '10

Hi, I am running FreeSWITCH Version 1.0.trunk (17134). I just got my spa3102 and have configured my phone as extension 1000. There's no PSTN or external VoIP yet. I am getting a lag whenever I dial out from the phone. For eg. when I dial the voicemail(4000) I hear "please enter your id....." not the part before that. There's a 10 sec lag when I call the demo IVR, there's also a lag when I dial other extensions(softphones). There's no lag whatsoever while dialing from softphones. Please recommend where I should be looking to resolve this before I move on to more adventurous stuff :) Thanks gs

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How does Freeswitch CDR determine which gateway was used in failover?
by Frank Church 30 Apr '10

30 Apr '10

How does Freeswitch CDR determine which gateway was used in failover? I am looking at the xml cdr and cdr_csv and they don't appear to show which gateway was used in the bridge data. Is there a separate uuid for each bridging attempt, that can be obtained from the CDR, or the event socket as last resort? -- Frank Church ======================= http://devblog.brahmancreations.com

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user_exists / user_registered?
by Phillip Jones 30 Apr '10

30 Apr '10

Hi there, You can call the api command user_exists - to see it the user is defined in the /directory. Is there an api command that tells you whether a user is currently registered? Like user_registered?? Thanks Pj

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Checking whether both RTP-streams are up or not
by Helmut Kuper 30 Apr '10

30 Apr '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I'm looking for an (easy) way to detect whether both RTP legs to caller resp. callee are streaming. Just for debugging reasons. Because sometimes there is only one RTP stream working, e.g. callee can hear caller but not vice versa. Is there already a solution or method in FS to debug this? best regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFL2tnw4tZeNddg3dwRAjbaAKCEjNysCFYKkSoyo/PrE0r50xHPUQCcCoT1 ybWe6dovDrsayyFnlwKWFeI= =lcFu -----END PGP SIGNATURE-----

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No PRI Call Until After Manual "load mod_openzap"
by Jerry Richards 30 Apr '10

30 Apr '10

I upgraded to FS snapshot as of 04-27-2010 and Wanpipe 3.5.11 and now, for some reason, I cannot make an outbound call unless I manual execute "load mod_openzap". Anyone know why? I posted the initialization, failed call attempt, load mod_openzap, and successful call at http://pastebin.freeswitch.org/12845. Thanks And Best Regards, Jerry

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Good book on (VoIP) telephony?
by GillesToo 30 Apr '10

30 Apr '10

Hello I was wondering: For those of you experienced Freeswitch users, is there a good book on telephony that you would recommend for software people who have never worked in the telephony business and would provide a good basis to make sense of VoIP in general, and Freeswitch in particular? FWIW, here's what Amazon returns with "telephony": www.amazon.com/s/qid=1272412320/ref=sr_pg_2?keywords=telephony&rh=n:!1000,i… Thank you. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Good-book-on-VoIP-telephony-t… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Difference between effective-caller-id and outbound-caller-id
by Frank Church 30 Apr '10

30 Apr '10

What are the differences between the two and when is one more appropriate than the other? What are the use cases? -- Frank Church ======================= http://devblog.brahmancreations.com

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FS Cluster
by Phillip Jones 29 Apr '10

29 Apr '10

Hi there, I am sure this has been asked before but I can not find any reference to the subject in the wiki. Basically - I have two FreeSWITCH servers, FS1 and FS2. Both sit behind OpenSIPS for load balancing Phone A is registered on FS1. A call comes in on FS2 for phone A. How do I get that call across to FS1. So far I have the internal profile of FS1 and FS2 pointing to the same ODBC database. What else do I have to do? If anyone can point me in the right direction I would be grateful. Thanks Pj

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Dial Timeout Issue
by Peder 29 Apr '10

29 Apr '10

I'm having an issue where originate_timeout doesn't seem to be working as I think it would. I am using lcr to get the gateways. I can bridge to either gateway1 or 2 just fine. Then I set it up with bridge using both gateways for failover and it worked fine. I decided to change gateway1 to a fake IP to test failover and it works, but it takes 30 seconds to timeout. I then found originate_timeout and set it to 4 seconds, but it doesn't timeout. It still takes 30 seconds. Here is the snippet from my dialplan: <action application="set" data="originate_timeout=4"/> <action application="bridge" data="${lcr_route_1}|${lcr_route_2}"/> Here is the debug that shows it is being set: EXECUTE sofia/internal/1111(a)192.168.1.108 set(originate_timeout=4) 2010-04-28 16:06:28.265187 [DEBUG] mod_dptools.c:818 sofia/internal/1111(a)192.168.1.108 SET [originate_timeout]=[4] EXECUTE sofia/internal/1111(a)192.168.1.108 bridge([lcr_carrier=tex,lcr_rate=0.01900]sofia/gateway/Tex/5551212|[lcr_carr ier=tel,lcr_rate=0.03000]sofia/gateway/Tel/5551212) Then I get this: 2010-04-28 16:07:00.273156 [NOTICE] sofia.c:4547 Hangup sofia/external/5551212 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE] And it successfully goes out gateway2, but it is 30 seconds later, not 4 as I would guess. Am I using the timeout wrong? Or is there some other setting I need instead?

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