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INVALID_NUMBER_FORMAT error
by Dhairya Vora 05 Dec '11

05 Dec '11

Hi group, I am getting this error while generating outbound calls using freeswitch. *freeswitch@server> 2011-12-05 22:01:44.760933 [ERR] mod_sofia.c:4279 Invalid URL 2011-12-05 22:01:44.760933 [NOTICE] mod_sofia.c:4660 Close Channel N/A [CS_NEW] 2011-12-05 22:01:44.760933 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2011-12-05 22:03:50.208776 [ERR] mod_sofia.c:4279 Invalid URL* Can anyone help me with this? -Dhairya

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Softphones could not call each other but could register on the FS server
by ayobami 05 Dec '11

05 Dec '11

I just git pulled the Freeswitch source code and compiled, everything went well, I mean the compilation, but the problem that arose now is sip phones could not call each other again, now the soft phones could register on the FS server because I could use them to call 5000, 4000 and some of my custom numbers, but when I tried to call an extension from another extension, the operator keep saying that the extension is not available, I have troubleshooted and dont know what to do, please people help me out. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Softphones-could-not-call-eac… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] mod_callcenter ringback - help
by Jeferson Rodrigo Almeida 05 Dec '11

05 Dec '11

> > Hi... > > I'm using mod_callcenter > > I need to send a ringback (or any other sound, different of the default > moh) to the caller when a agent phone is ringing, and I don't know how to > do it. > > Basically, I need to playback other sound to the caller when > agent-offering event is fired. > > Is there any way to configure a dialplan after the call is bridged to the > agent? > Is there any way to modify the moh for this call in real-time? (I can > capture the event...) > > Any help will be appreciated. > > Ps.: Sorry for any mistake in my "school" english. > > Thanks > > -- > Jeferson Rodrigo Almeida > Engenheiro de Computação > jefersonparanaense(a)gmail.com

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Almost there... Dingaling help please
by Jacob Smith 05 Dec '11

05 Dec '11

At this point, I can make calls using Dingaling through Google Voice. The problem is receiving: the phone rings, but when I answer there is silence and after a few seconds, disconnect. Here is a log from one attempt: http://pastebin.freeswitch.org/17937 The Jingle profile and dialplan is just like http://wiki.freeswitch.org/wiki/Google_Voice Hoping that this is an easy fix, I have one more question. How can I answer the call with an IVR instead of sending the call straight to an extension? The only forwarding option I can find is in the Jingle profile where you specify the extension. Thank you, Jacob

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play and get digits with dynamic conference
by Rodney 05 Dec '11

05 Dec '11

I have tried the nb_conference example in the default.xml but I think I am doing something wrong. when i transfer to that extension it drops me into a single static conference of the variable . what do i need to add or change to prompt my callers transfered to the nb_conference to enter a 4 digit conference number without pin that gives them the same options as my static conferences on the default profile? I know play and get digits must be involved but would appreciate the help. thanks. <extension name="nb_conferences"> <condition field="destination_number" expression="^\*(30\d{2})$"> <action application="answer"/> <action application="conference" data="$1-${domain_name}@default"/> </condition> </extension>

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Segmentation fault (core dumped)
by Miha Zoubek 05 Dec '11

05 Dec '11

Hi, I have added this in mod_radius_cdr.c and run make&make install. if (channel){ const char *no_radius_start = switch_channel_get_variable(channel, "no_radius_start"); if (switch_true(no_radius_start)){ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "[mod_radius_cdr] Skipping Radius Start\n"); return SWITCH_STATUS_SUCCESS; } } and for radius_stop. if (channel){ const char *no_radius_stop = switch_channel_get_variable(channel, "no_radius_stop"); if (switch_true(no_radius_stop)){ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "[mod_radius_cdr] Skipping Radius Stop\n"); return SWITCH_STATUS_SUCCESS; } } When I run call a get this: Segmentation fault (core dumped) Do you know why I am getting this? Regrdas, Miha

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get cepstral volume lower
by Madovsky 05 Dec '11

05 Dec '11

I'm trying the SSML example from mod_cepstral wiki <action application="speak" data="cepstral|david|<prosody volume=13>Thank you for holding</prosody>"/> but the voice doesn't say the sentence but says something like "slash blablabla...". Any idea why I can't embed SSML tag in the cepstral sentence ? Thanks

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acl validation
by Darcy 05 Dec '11

05 Dec '11

Looking through my call log, I see the following item in the log, this IP is not in the acl Domain List, it is not a profile and it is not registered on the switch, Is there something I am missing on this that needs to be set. These guys are attempting calls to Jamaica. 2011-10-31 09:11:31.801420 [DEBUG] sofia.c:5798 IP 91.212.226.30 Approved by acl "domains[]". Access Granted. <list name="domains" default="deny"> <node type="allow" cidr="xxx.xxx.xx.xx/32"/> I have 5 more entries like so in this switch. etc. </list> Darcy Primrose

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"481 Call/Transaction Does Not Exist" when hanging up before call connects
by Charlie Orford 05 Dec '11

05 Dec '11

Hi list, When we make a call from an FS extension to a PSTN number (via our ITSP gateway provider) and hang-up before the call completes, FS replies to the CANCEL request with "481 Call/Transaction Does Not Exist" and the call continues to ring on the remote end. If we hangup after the call has connected, it works with no problem. I have looked through a SIP trace of this happening and to my (untrained eye) nothing seems obviously wrong (i.e. tag, call id and branch values all seem to be correct). I'm using the latest git snapshot from 2011-11-30 18-14-24 -0600. For a sip trace showing the problem, see: http://pastebin.freeswitch.org/17908 For a sip trace showing a successful hangup, see: http://pastebin.freeswitch.org/17909 *Note: ip addresses, domain and called number have been altered for privacy. Any help or insight is much appreciated. Kind Regards, Charlie

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mod_rtmp: "Read error" when attempting calls from FS to RTMP client.
by peely 03 Dec '11

03 Dec '11

Hi, I have a strange issue when I attempt to bridge a call from FS out to an RTMP client. As soon as the call is offered up to the client, the call is disconnected immediately i.e at the "ringing" stage. The strange thing is that as soon as I issue an "fsctl loglevel 9" then everything is fine. I don't know whether this is because of any additional load, as nothing above "ERR" is printed in each case. When the call is cleared immediately then I am getting "[ERR] rtmp.c:713 Read error" in the logs. mod_rtmp is compiled from today's git. Is there a possibility something is trying to happen too quickly in the handshake before data has been received from the client? Outbound calls from client to FS work flawlessly and incoming calls too, so long as I seem to have debug logging enabled. I'm not sure if it's relevant, but I'm bridging to the UUID of the session AKA "<action application="bridge" data="rtmp/cd421c76-1614-11e1-be92-397d873fd9b8"/>" Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-rtmp-Read-error-when-atte… Sent from the freeswitch-users mailing list archive at Nabble.com.

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