- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSwitch - mod_skypopen alsa driver on centos 64 bit, x86_64
by Shamun toha md 25 Sep '10

25 Sep '10

Hello, Alsa driver is installed, but i have no sound. I was trying to use alsamixer or etc. But nothing is just allowing me. ex: http://gist.github.com/raw/596026/41aff3b9d8fe24492cfa8fab57ecaa3da02e69f7/… Thanks for guiding..

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No caller ID sent to a bridged outside call
by Mario 25 Sep '10

25 Sep '10

I am bridging a call from either inside or outside to an emergency number on an att cellphone which works fine. I am trying to pass the original caller id or anything else for that matter as the caller id/name. I only get "unknown". I also tried to send to a VOIP number and it also gets "unknown" so I am thinking it is not att. I tried various combinations of things I found on the web but nogo. Anyone see the problem below? Thanks! Mario <extension name="mx911"> <condition field="destination_number" expression="^2911$"> <action application="set" data="effective_caller_id_name=Emergency"/> <action application="set" data="effective_caller_id_number=9161234567"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/uid1/19161112222"/> </condition> </extension>

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[OT] Own Posts
by Ross McKillop 25 Sep '10

25 Sep '10

I don't receive my own posts to this list, despite enabling it on the Mailman control panel ? Is it disabled globally or am I doing something wrong? Ross

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Call Forwarding Problem
by Mike 24 Sep '10

24 Sep '10

Hi, Here's a problem I posted about on the IRC some time ago - I said I'd do some traces and put them in pastebin, which I've done here: http://pastebin.freeswitch.org/13970 A brief description of the problem and the calls involved in the pastebin. Incoming calls are delivered to the Freeswitch from a SIP trunking provider who are also used to route outgoing calls. This is not via a gateway. The SIP trunking provider insists that for outgoing calls, the data in the "From" field in the INVITE message contains a PSTN number that they have 'associated' with the SIP trunking service, e.g. From: "Zoltan" <sip:02031375892@79.125.110.116> is OK. From: "Zoltan" <sip:anonymous@79.125.110.116> would be rejected by them, as would From: "Zoltan" <sip:01344299800@79.125.110.116> as 01344299800 is 'someone elses' number, not one of theirs that they have allocated to us. The first call in the pastebin shows a Polycom phone making an outgoing call that works / is allowed as the effective_caller_id_number is set to an "allowed" number 02031375892. The second call, starting at line 871 shows a call originating in the PSTN from number 01344299800 to the Polycom. The SIP provider delivers the call to the FreeSwitch which in turn sends the call to the Polycom. The Polycom has been set to divert all incoming calls to number 07770380800 and replies with a "SIP/2.0 302 Moved Temporarily" with the forwarded to number as the contact. This in turn causes the FreeSwitch to generate an INVITE message which is sent to the trunk provider with: From: "01344299800" <sip:01344299800@79.125.110.116> and To: <sip:07770380800@79.125.4.174> Since 01344299800 is not an "allowed" number according to the provider, the call is rejected. Putting aside the argument for now of whether the SIP provider should be carrying out this rather brutal form of "CLI screening" (the data in the 'From' field appears as the A Party Number / CLI once the call hits the PSTN / mobile networks), is there any way to modify the behaviour of this call so that in response to the "SIP/2.0 302 Moved Temporarily" message, the INVITE which is sent out has the number from the effective_caller_id_number variable for the forwarding extension, rather than the number from the "From" field of the incoming invite? Hope this makes sense, Mike

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mod_callcenter dropping call
by Troy Anderson 24 Sep '10

24 Sep '10

Hi, I'm trying to play around with mod_callcenter. The wiki is well written and thorough, except that I can't get it to work ;) Hopefully I'm just missing something simple. I have it set up with the sample callcenter.conf.xml from a fresh install. I added this to my default dialplan: <extension name="callcenter test"> <condition field="destination_number" expression="^11$" break="on-true"> <action application="answer"/> <action application="playback" data="ivr/ivr-hello.wav"/> <action application="callcenter" data="support@default"/> <action application="playback" data="misc/error.wav"/> </condition> </extension> When I dial 11 from a local phone, I hear "hello" and then "error" and in fs_cli, with logging set to 7, there is not one line of output related to callcenter except for the fact that it executed the above extension. It's as if the callcenter application never got executed. However, I can add and remove members, tiers, and agents to using callcenter_config. Is this the correct way to drop a call into a callcenter queue? Thanks for any help! Troy

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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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