[Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call

I have an inbound call via OpenZap, when I attempt to bridge to a SIP extension, I get the ring tone (inbound line) up until the bridge fails (for timeout or do not disturb). At this point the call is answered and then my dial plan moves on to attempt another bridge to different extensions. So

Re: [Freeswitch-users] Announcing mod_easyroute

this is a killer most likely for sw billing developers. another breakthrough... On Wed, Jan 14, 2009 at 10:52 AM, William Suffill wrote: > Looks like a solid contribution as always Ken. I agree it should be an > interesting year with the way things are shaping up. > Glad to see someone pointing

Re: [Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call (partial call log added)

> I have an inbound call via OpenZap, when I attempt to bridge to a SIP > extension, I get the ring tone (inbound line) up until the bridge fails > (for timeout or do not disturb). At this point the call is answered and > then my dial plan moves on to attempt another bridge to different > exte

Re: [Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call

Have a look here: http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones On Wed, Jan 14, 2009 at 4:03 AM, Scott Ellis wrote: > I have an inbound call via OpenZap, when I attempt to bridge to a SIP > extension, I get the ring tone (inbound line) up until the bridge fails > (for timeout or do not

[Freeswitch-users] No caller/called ID received (Wildcard X101P)

Hi all, I'm a new FreeSwitch user and this is my first email to the list. I'm trying to configure my Home PBX with a Wildcard X101P (configured as FXO) and I have a problem receiving the caller/called ID from PSTN. This is the content of file "openzap.conf": [span zt] name => OpenZAP number =>

Re: [Freeswitch-users] No caller/called ID received (Wildcard X101P)

number => 1 This value should be set to the DID of the FXO line. That way when a call hits FS it will go to that extension in the dialplan. This is unrelated to callerid, it's the destination not the source. If the line has caller-id it will also be available when it's collected after the 2nd rin

Re: [Freeswitch-users] No caller/called ID received (Wildcard X101P)

I noticed tonegroup=es. What country are you in and do you know what method they use to do dtmf. Most likely we need a small tweak to set the dtmf method for your country. Mike On Jan 14, 2009, at 9:05 AM, Anthony Minessale wrote: number => 1 This value should be set to the DID of the

[Freeswitch-users] NORMAL_TEMPORARY_FAILURE problem

Hi, I want to know exactly what does this hangup_cause means: "NORMAL_TEMPORARY_FAILURE". I'm receiving lots of those. Is the SIP provider to blame, or is my setup? I took a look at the sip communication (Wireshark/tcpdump), and I couldn't find a response from the sip provider that matches. All

Re: [Freeswitch-users] NORMAL_TEMPORARY_FAILURE problem

Sip cause code to Q.850 cause code translations can be found in RFC4497 section 8.4.4. FreeSWITCH uses Q.850 codes internally so you will typically see those in the logs. We do pass the sip cause codes across a sip to sip bridge. Mike On Jan 14, 2009, at 9:18 AM, Alexandru Nedelcu wrote:

Re: [Freeswitch-users] No caller/called ID received (Wildcard X101P)

Hi, Anthony, I think that's my problem, when I receive a call from the PSTN, FS receive number 1 instead of my house number and I don't know why. Michael, I live in Spain, Is it not "es" the tonegroup I should use? Thank you very much. On Wed, Jan 14, 2009 at 3:18 PM, Michael Jerris wrote: >

Re: [Freeswitch-users] Announcing mod_easyroute

Excellent E ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

Re: [Freeswitch-users] No caller/called ID received (Wildcard X101P)

like i said, in openzap.conf change number => 1 to number => and you will not see the call arriving at ext 1 anymore this has nothing to do with caller id. We so far have only tested the caller id code in US so if es uses the same one as uk or jp we may have to add some more code. Can you pr

[Freeswitch-users] ipauth - directory

I noticed an "ip=" setting in the brian.xml sample file. The comments state that this is used for ipauth (IP based authentication?) What exactly is this setting. I cannot find anything in the wiki about it. Does it replace the use of the + ACL mechanism for IP authentication? -- -

Re: [Freeswitch-users] ipauth - directory

cidr= and the domains acl in acl.conf.xml then apply that ACL to the sofia profile. /b On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: > I noticed an "ip=" setting in the brian.xml sample file. > The comments state that this is used for ipauth (IP based > authentication?) > > Wha

Re: [Freeswitch-users] ipauth - directory

Yes I know that. But what does the "ip=" setting do? Brian West wrote: cidr= and the domains acl in acl.conf.xml then apply that ACL to the sofia profile. /b On Jan 14, 2009, at 9:15 AM, Apostolos Pantsiopoulos wrote: I noticed an "ip=" setting in the brian.xml sample file. The comments

Re: [Freeswitch-users] No caller/called ID received (Wildcard X101P)

Tomás wrote: > Hi, > > Anthony, I think that's my problem, when I receive a call from the PSTN, > FS receive number 1 instead of my house number and I don't know why. If you use SIP trunking or something like an ISDN-PRI line, the number the call is to is delivered as part of the signaling, which

[Freeswitch-users] mod_sofia: NAT-ping & RPID bounties

Hi all, I have just post two bounties (NAT-ping and RPID changes). So I hope someone may be interested :-) BTW: Both changes have to be (and stay in the future) integral part of FreeSWITCH code. Best regards, kokoska.rokoska ___ Freeswitch-users m

[Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV]

After a time I receive the following error when a call comes in on our OpenZap span 2: parse error [-3012] [Q931E_INVALID_CRV] Here's the log 2009-01-14 13:14:11 [DEBUG] ozmod_isdn.c:320 zap_isdn_931_34() Yay I got an event! Type:[4d] Size:[103] CRV: 23 (0x17, CTX: Originator) 2009-01-14 13:14:11

Re: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV]

Some more info: Now I cannot get any incoming call through OpenZAP. "oz dump 1" and "oz dump 2" show that all channels are DOWN. After reload of mod_openzap everythings works again. I receive the following messages for each incoming call: 2009-01-14 19:07:29 [DEBUG] ozmod_isdn.c:777 zap_isdn_921_

[Freeswitch-users] Internal vs. External Call, How to detect answered call?

Hi, I'm unable to detect whether a call was answered or not. Using a custom Java API to connect to Freeswitch Socket Interface, i observed that: 1) session.originate("sofia/internal/1003%192.168.50.94") does block until the callee picks up while 2) session.originate("sofia/gateway/sip.gafachi.

[Freeswitch-users] SVN Commits now tweet!

http://twitter.com/freeswitchsvn Check it out. ;) /b ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswi

Re: [Freeswitch-users] OpenZAP parse error [-3012] [Q931E_INVALID_CRV]

I believe these are all symptoms of something that Stefan is working on: better Q931 timers. It's been on the todo list for some time but we've had absolutely NOBODY willing to pony up serious $$ to support OpenZAP development which means it is progressing at the speed of developers' free time. -M

[Freeswitch-users] Registrar performance

Hi all, could someone be so kind and share some data about concurrently registered users at FreeSWITCH from production usage? I have done few sipp testing and fall in doubt... BTW: I use xml_curl for directory and the data are served through: 1. apache, php, mysql 2. lighttpd, "c" binary, mysql

[Freeswitch-users] ilbc & alaw transcoding not working correctly?

Hello all!, I am using freeswitch with Nokia E65 phone. When phone uses iLBC codec and I am trying "monkey demo" in the example IVR sound is choppy. When using alaw/ulaw there is no problem, it works OK. IVR prompts work OK with G711 and iLBC, there is no problem. So probably freeswitch is not

[Freeswitch-users] Using mod_managed Linux/Mono 2.02

Has anyone had any luck using mod_managed under linux with mono yet? The Wiki looks to still be lacking some linux installation instructions. I feel like I'm close but missing something simple. I got as far as adding "languages/mod_managed" to the /usr/src/freeswitch-1.0.2/modules.conf withou

Re: [Freeswitch-users] ilbc & alaw transcoding not working correctly?

Please collect the sip traces pcaps and details and open a jira. /b On Jan 14, 2009, at 4:51 PM, Hostinsky Miroslav wrote: Hello all!, I am using freeswitch with Nokia E65 phone. When phone uses iLBC codec and I am trying "monkey demo" in the example IVR sound is choppy. When using alaw/u

Re: [Freeswitch-users] Failed bridge in dial plan still answers an OpenZap inbound call

Thanks Anthony, got to say I am hugely impressed with the software - I am another Asterisk refugee :-) So the answering of the call even though the bridge fails is correct operation for the system? (Just curious) Scott Anthony Minessale wrote: Have a look here: http://wiki.freeswitch.or

[Freeswitch-users] mod_managed failing to load on CentOS 5.2

Got mod_managed compiled and installed. Now it isn't loading. See below... 1) Donwloaded fresh from SVN 2) Compiled... and installed.. OK [r...@phone2 mod_managed]# export PKG_CONFIG_PATH=/usr/local/lib/pkgconfig [r...@phone2 mod_managed]# make [r...@phone2 mod_managed]# make install 3) Added

Re: [Freeswitch-users] Using mod_managed Linux/Mono 2.02

The managed assembly should be the same on both platforms. The correct name is FreeSWITCH.Managed.dll. I'll get a patch to the mod_managed/managed/Makefile. Meanwhile, simply renaming mod_managed_lib.dll should work. After that, make sure there's a "managed" subdirectory where the modules are.

[Freeswitch-users] zapata.conf immediate=yes in Asterisk - Freeswitch equivalent?

Searched the wiki and mailing lists as best I can, but with no luck. How do I get OpenZap to answer a call immediately? (I do not need caller id) Scott ___ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.or

[Freeswitch-users] Country specific tones - how to contribute?

I have tracked down a set of au tones from the mailing list, which I am going to verify. How do I go about getting these added into the default build so that they are available for all in future? I tried and this did not work - where does it try and load the ring tone from? I have entries in

Re: [Freeswitch-users] Country specific tones - how to contribute?

Scott Ellis wrote: > I have tracked down a set of au tones from the mailing list, which I am > going to verify. How do I go about getting these added into the default > build so that they are available for all in future? Maybe by posting a patch to the bug tracking system or the development lis

[Freeswitch-users] Changes in PlayAndGetDigits

Hi, Is there a change in the playAndGetDigits api? In the old release, 11102, my lua script is working but is not working in the latest release. The error I am getting is " Error in playAndGetDigits expected 10..10 args, got 9 ". Thanks, JB ___ Freesw