[Freeswitch-users] How to get digitals and stop play when speak tts? Just like session:playAndGetDigits do
Hi all, I'm writing lua ivr scirpt, Does some known how to get digitals and stop play when speak tts? Just like session:playAndGetDigits do. Thanks lots! Best Regards! -- Lei.Tang lei.tl...@gmail.com ___ 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] mod_valet_parking: auto reports on wrong leg of call
Brian Stafford wrote: Brian West wrote: You have to be doing it wrong then. Can you show us your dialplan you should have two extensions one for the lot range and one to attended transfer someone into the lot. /b The relevant excerpt from the dialplan is extension name=valet_unpark condition field=destination_number expression=^(41[0-9])$ action application=answer/ action application=valet_park data=valet_lot $1/ /condition /extension extension name=valet_park condition field=destination_number expression=^(420)$ action application=answer/ action application=valet_park data=valet_lot auto in 410 419/ /condition /extension x410-419 are the slots and 420 parks a call. Parking by picking one of 410-419 works fine and subsequently dialling them from another works fine, I added x420 for the auto feature. Regards Brian _ Any clues what I'm doing wrong? Is more information needed? Brian ___ 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
[Freeswitch-users] Users hanged up for unknown reason
Hi, I have a strange problem. I control FS with commands sent by tcp in response to events published via tcp. I do something like: 1) call 1st user 2) call 2nd user 3) 1st and 2nd talk 4) call another user 5) 1st and another talk etc... Sometimes (quite regularly) users are hanged up (with cause NORMAL_CLEARING) even if they do not hangup manually. I pasted one such scenario in pastebin (http://pastebin.freeswitch.org/10955), it includes logs from commands sent by me and events received from FS. Could someone take a look and see what am I doing wrong? The scenario includes 3 users 1st user (Unique-ID: f076261a-4537-40f2-b46d-933141320314) is supposed to be connected all the time but gets diconnected 2nd user (Unique-ID: ebdfb398-ec82-4760-9f79-81364e0f37b6) is supposed to talk for a few seconds and get killed 3rd user (Unique-ID: d5cd839e-793c-4b3c-adda-327841672a5f) is supposed to work like 2nd user All of them are simulated by dialplan extensions (using answer and playback tools), but the same thing happends for xlite or cisco phone. Maciej Aniserowicz -- View this message in context: http://n2.nabble.com/Users-hanged-up-for-unknown-reason-tp3937601p3937601.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ 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
[Freeswitch-users] Precompiled Windows Binaries
Hi, I have read the Docs on the Wiki (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) but am still not sure of what the different Windows install files are. Currently, the Windows Installer directory contains :- LATEST_SVN_15106 - 6 Bytes freeswitch-1.0.4.exe - 42 Megabytes freeswitch.exe - 32 Megabytes I have installed the freeswitch-1.0.4.exe file which is dated 3rd September. The freeswitch.exe file is dated 7th October and think that it contains the minor updates since 3rd September ? Could someone who knows FreeSwitch under windows help me understand the two files please ? I chickened out of running the later exe in case it did something to the running install of FreeSwitch 1.0.4, is it safe to run the newer exe with the old one already installed ? What will it actually do ? regards Dave___ 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] Precompiled Windows Binaries
Hi Dave, These are supported by Carlos Talbot . They also include Freepbx v3 Just as you said freeswitch-1.0.4.exe is the tagged release and freeswitch.exe is a newer svn snapshot. There should be no problems installing the new version allthough best to just try and see! Not sure why the newest one is from October 7th. Jeff Dave Stevenson wrote: Hi, I have read the Docs on the Wiki (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) but am still not sure of what the different Windows install files are. Currently, the Windows Installer directory contains :- LATEST_SVN_15106 - 6 Bytes freeswitch-1.0.4.exe - 42 Megabytes freeswitch.exe - 32 Megabytes I have installed the freeswitch-1.0.4.exe file which is dated 3rd September. The freeswitch.exe file is dated 7th October and think that it contains the minor updates since 3rd September ? Could someone who knows FreeSwitch under windows help me understand the two files please ? I chickened out of running the later exe in case it did something to the running install of FreeSwitch 1.0.4, is it safe to run the newer exe with the old one already installed ? What will it actually do ? regards Dave ___ 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 -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ 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] Precompiled Windows Binaries
Jeff, thanks a lot for the reply. I was a little confused by the fact that the SVN Snapshot was some 10MB smaller than the Full 1.0.4 file so worried that I might lose something. As you say though, think that I'll cross my fingers and try the updated release. I am running FreeSwitch on a test machine at the moment until the target hardware arrives - hopefully tomorrow, so I can afford to have a little play. You mentioned FreePBX V3. I had been fumbling around trying to work out what this is and from what I've read, it seems to provide a GUI Front End for configuring FreeSwitch ? I am guessing that while it has been installed with FreeSwitch, I then need to run the FreePBX Installer to update the FreePBX/FreeSwitch configuration on my hardware ? When I start FreeSwitch, it does not automatically load the WAMPServer. When I start WAMPServer manually, and open up localhost (127.0.0.1) in a web browser, I can see the WampServer logo and various tools such as phpinfo() and phpmyadmin. FreePBX is there under Your Projects. When I opened this up the first time, it appeared to want to install FreePBX over FreeSwitch, I tried to abort this when it was going to overwrite some FreeSwitch conf files and I thought I'd better not go on until I had a better idea what was happening. I backed out of the FreePBX install and now I can't get the FreePBX or phpmyadmin pages up again (missing files) so it looks like I'm going to have to reinstall anyway. So, for next time,am I right in thinking that I should proceed with running the FreePBX install from the WAMPServer menu ? regards Dave - Original Message - From: Jeff Lenk jl...@frontiernet.net To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, November 03, 2009 2:48 PM Subject: Re: [Freeswitch-users] Precompiled Windows Binaries Hi Dave, These are supported by Carlos Talbot . They also include Freepbx v3 Just as you said freeswitch-1.0.4.exe is the tagged release and freeswitch.exe is a newer svn snapshot. There should be no problems installing the new version allthough best to just try and see! Not sure why the newest one is from October 7th. Jeff Dave Stevenson wrote: Hi, I have read the Docs on the Wiki (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) but am still not sure of what the different Windows install files are. Currently, the Windows Installer directory contains :- LATEST_SVN_15106 - 6 Bytes freeswitch-1.0.4.exe - 42 Megabytes freeswitch.exe - 32 Megabytes I have installed the freeswitch-1.0.4.exe file which is dated 3rd September. The freeswitch.exe file is dated 7th October and think that it contains the minor updates since 3rd September ? Could someone who knows FreeSwitch under windows help me understand the two files please ? I chickened out of running the later exe in case it did something to the running install of FreeSwitch 1.0.4, is it safe to run the newer exe with the old one already installed ? What will it actually do ? regards Dave ___ 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 -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ 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 ___ 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] SIP Overlap support?
hi anthony, i believe, that there is no problem with the communication between fs and the cirpack (everything works to smooth as if this could be possible). if fs sends the 484, the cirpack sends more digits to fs (if there are some), so this works as it should. the problem is, that fs ends the session/socket after a 484, so that the cirpack sends the following digits into another socket. you wrote about a 1 line patch, which might not have been implemented - at least it seems so. is there a way to get someone of the sofia devs to fix this small problem, so that fs sends the 484 without ending the session/socket and waiting for an answer of the cirpack? we would take care of the rest. kind regards, dennis 2009/10/15 Anthony Minessale anthony.miness...@gmail.com: right you can reply 484 in your dp at any time action application=respond data=484 Address Incomplete/ then it should try again. The bit i can't remember is if we committed a certain 1 line patch that makes sofia parse the next invite to the same call properly, the patch was to the sofia lib itself so test it and see. I may need to dig up the answer again from the sofia dev. ___ 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
[Freeswitch-users] portaudio error
Hello
Debian lenny with svn15321
freeswi...@internal load mod_portaudio
-ERR [module load file routine returned an error]
2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input
devicefreeswi...@internal 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974
Cannot find an input device
2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading
module /opt/freeswitch/mod/mod_portaudio.so
**Module load routine returned an error**
configuration name=portaudio.conf description=Soundcard Endpoint
settings
!-- indev, outdev, ringdev:
partial case sensitive string match on something in the name
or the device number prefixed with # eg #1 (or blank for default) --
!-- device to use for input --
param name=indev value=1/
!-- device to use for output --
param name=outdev value=1/
!--device to use for inbound ring --
param name=ringdev value=1/
!--File to play as the ring sound --
!--param name=ring-file value=/sounds/ring.wav/--
!--Number of seconds to pause between rings --
!--param name=ring-interval value=5/--
!--Enable or Disable dual_streams--
!--param name=dual-streams value=true/--
!--file to play when calls are on hold--
param name=hold-file value=$${hold_music}/
!--Timer to use for hold music (i'd leave this one commented)--
!--param name=timer-name value=soft/--
!--Default dialplan and caller-id info --
param name=dialplan value=XML/
param name=cid-name value=$${outbound_caller_name}/
param name=cid-num value=$${outbound_caller_id}/
!--audio sample rate and interval --
param name=sample-rate value=48000/
param name=codec-ms value=10/
/settings
/configuration
Any help would be appreciated.
--FC
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Re: [Freeswitch-users] SIP Overlap support?
The patch was it's ability to accept subsequent invites. Your problem is that in sip each new attempt to send an invite is another call. 484 is a final response so the call with too few digits is terminated. On Tue, Nov 3, 2009 at 9:57 AM, Dennis oderm...@googlemail.com wrote: hi anthony, i believe, that there is no problem with the communication between fs and the cirpack (everything works to smooth as if this could be possible). if fs sends the 484, the cirpack sends more digits to fs (if there are some), so this works as it should. the problem is, that fs ends the session/socket after a 484, so that the cirpack sends the following digits into another socket. you wrote about a 1 line patch, which might not have been implemented - at least it seems so. is there a way to get someone of the sofia devs to fix this small problem, so that fs sends the 484 without ending the session/socket and waiting for an answer of the cirpack? we would take care of the rest. kind regards, dennis 2009/10/15 Anthony Minessale anthony.miness...@gmail.com: right you can reply 484 in your dp at any time action application=respond data=484 Address Incomplete/ then it should try again. The bit i can't remember is if we committed a certain 1 line patch that makes sofia parse the next invite to the same call properly, the patch was to the sofia lib itself so test it and see. I may need to dig up the answer again from the sofia dev. ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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] Sipura Codec Problem
I think you can edit the prefs in your sipura and change it to the correct string. On Tue, Nov 3, 2009 at 11:47 AM, Mariano de Llano mariano.dell...@gmail.com wrote: Hi, I'm having a problem with a Sipura, it is sending for the G729 the tag G729a witch is not correct due the RFC. Media Attribute (a): rtpmap:18 G729a/8000 FS is returning (200OK) Media Attribute (a): rtpmap:96 G729/8000 I think that the problem is that FS is not matching the codec, so it returns the first dynamic payload which is 96. I think that I've seen post with a similar issue, and the solution was to change the tag before it hit the switch, so, what I've done is to change the switch_r_sdp (I have the rest of the parameters correct due I also use it to dynamically change the codecs order) and it's changing the SDP, but when FS sends the 200OK it is returning to the endpoint: Media Attribute (a): rtpmap:96 G729/8000 Which is exactly the same problem that I have without the transformation of the SDP. Is it correct? Do I have another solution? Thanks ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
[Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch
Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave___ 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] Sipura Codec Problem
FIx your sipura to NOT include the a in the codec its in the admin section of the UI on the ATA. /b On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote: Hi, I'm having a problem with a Sipura, it is sending for the G729 the tag G729a witch is not correct due the RFC. Media Attribute (a): rtpmap:18 G729a/8000 FS is returning (200OK) Media Attribute (a): rtpmap:96 G729/8000 I think that the problem is that FS is not matching the codec, so it returns the first dynamic payload which is 96. I think that I've seen post with a similar issue, and the solution was to change the tag before it hit the switch, so, what I've done is to change the switch_r_sdp (I have the rest of the parameters correct due I also use it to dynamically change the codecs order) and it's changing the SDP, but when FS sends the 200OK it is returning to the endpoint: Media Attribute (a): rtpmap:96 G729/8000 Which is exactly the same problem that I have without the transformation of the SDP. Is it correct? Do I have another solution? Thanks ___ 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] Sipura Codec Problem
Yes, that was my first option, but there many endpoints that I'm not able to configure. Basically it's a broadband solution where I have like 1000 endpoints that are out of my provisioning. Thanks, M On 03/11/2009, at 14:58, Brian West wrote: FIx your sipura to NOT include the a in the codec its in the admin section of the UI on the ATA. /b On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote: Hi, I'm having a problem with a Sipura, it is sending for the G729 the tag G729a witch is not correct due the RFC. Media Attribute (a): rtpmap:18 G729a/8000 FS is returning (200OK) Media Attribute (a): rtpmap:96 G729/8000 I think that the problem is that FS is not matching the codec, so it returns the first dynamic payload which is 96. I think that I've seen post with a similar issue, and the solution was to change the tag before it hit the switch, so, what I've done is to change the switch_r_sdp (I have the rest of the parameters correct due I also use it to dynamically change the codecs order) and it's changing the SDP, but when FS sends the 200OK it is returning to the endpoint: Media Attribute (a): rtpmap:96 G729/8000 Which is exactly the same problem that I have without the transformation of the SDP. Is it correct? Do I have another solution? Thanks ___ 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 ___ 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] Sipura Codec Problem
Actually, there were a few more misinterpretations in earlier software of Cisco Gateways, which RFC implementers had to address in RFC3551, strange ... RTP Payload Type 19 remains reserved because some implementations wrongly interpreted 13 decimal as 13 hexadecimal value. Another issue is G726 bit packing. Again some implementations used wrong bit packing and RFC3551 tried to partially resolve this conflict introducing new payload format named AAL2-G726 ... From: Brian West br...@freeswitch.org To: freeswitch-users@lists.freeswitch.org Sent: Tue, November 3, 2009 10:27:39 PM Subject: Re: [Freeswitch-users] Sipura Codec Problem Sounds like bad planning. I would send out a memo to your users and have them fix it. I have raised a bug multiple times with Cisco g729a is NOT valid. /b On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: Yes, that was my first option, but there many endpoints that I'm not able to configure. Basically it's a broadband solution where I have like 1000 endpoints that are out of my provisioning. Thanks, M ___ 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 ___ 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
[Freeswitch-users] Sipura Codec Problem
Hi, I'm having a problem with a Sipura, it is sending for the G729 the tag G729a witch is not correct due the RFC. Media Attribute (a): rtpmap:18 G729a/8000 FS is returning (200OK) Media Attribute (a): rtpmap:96 G729/8000 I think that the problem is that FS is not matching the codec, so it returns the first dynamic payload which is 96. I think that I've seen post with a similar issue, and the solution was to change the tag before it hit the switch, so, what I've done is to change the switch_r_sdp (I have the rest of the parameters correct due I also use it to dynamically change the codecs order) and it's changing the SDP, but when FS sends the 200OK it is returning to the endpoint: Media Attribute (a): rtpmap:96 G729/8000 Which is exactly the same problem that I have without the transformation of the SDP. Is it correct? Do I have another solution? Thanks ___ 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] portaudio error
On Tue, Nov 03, Andrew Thompson wrote: On Tue, Nov 03, 2009 at 12:01:10PM -0500, Frank Carmickle wrote: Hello Debian lenny with svn15321 freeswi...@internal load mod_portaudio -ERR [module load file routine returned an error] 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input devicefreeswi...@internal 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 Cannot find an input device 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_portaudio.so **Module load routine returned an error** Try installing the alsa development headers, it's got some stupid name on debian like libasound2-devel or something. Then re-build the portaudio module and library (a couple well placed make cleans should do it). Hi Libasound2-dev is still installed. I have had PA working in the passed. I think it was as of svn 14000 or so. Thanks for the help. --FC ___ 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] Sipura Codec Problem
so imagine how much money all those sipuras cost. They get all the money *and* have a bug and we are free and are supposed to break the rules for them. On Tue, Nov 3, 2009 at 12:11 PM, Mariano de Llano mariano.dell...@gmail.com wrote: Yes, that was my first option, but there many endpoints that I'm not able to configure. Basically it's a broadband solution where I have like 1000 endpoints that are out of my provisioning. Thanks, M On 03/11/2009, at 14:58, Brian West wrote: FIx your sipura to NOT include the a in the codec its in the admin section of the UI on the ATA. /b On Nov 3, 2009, at 11:47 AM, Mariano de Llano wrote: Hi, I'm having a problem with a Sipura, it is sending for the G729 the tag G729a witch is not correct due the RFC. Media Attribute (a): rtpmap:18 G729a/8000 FS is returning (200OK) Media Attribute (a): rtpmap:96 G729/8000 I think that the problem is that FS is not matching the codec, so it returns the first dynamic payload which is 96. I think that I've seen post with a similar issue, and the solution was to change the tag before it hit the switch, so, what I've done is to change the switch_r_sdp (I have the rest of the parameters correct due I also use it to dynamically change the codecs order) and it's changing the SDP, but when FS sends the 200OK it is returning to the endpoint: Media Attribute (a): rtpmap:96 G729/8000 Which is exactly the same problem that I have without the transformation of the SDP. Is it correct? Do I have another solution? Thanks ___ 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 ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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] Sipura Codec Problem
Sounds like bad planning. I would send out a memo to your users and have them fix it. I have raised a bug multiple times with Cisco g729a is NOT valid. /b On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: Yes, that was my first option, but there many endpoints that I'm not able to configure. Basically it's a broadband solution where I have like 1000 endpoints that are out of my provisioning. Thanks, M ___ 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] portaudio error
On Tue, Nov 03, 2009 at 12:01:10PM -0500, Frank Carmickle wrote: Hello Debian lenny with svn15321 freeswi...@internal load mod_portaudio -ERR [module load file routine returned an error] 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:964 Cannot find an input devicefreeswi...@internal 2009-11-03 11:56:47.047969 [ERR] mod_portaudio.c:974 Cannot find an input device 2009-11-03 11:56:47.047969 [CRIT] switch_loadable_module.c:871 Error Loading module /opt/freeswitch/mod/mod_portaudio.so **Module load routine returned an error** Try installing the alsa development headers, it's got some stupid name on debian like libasound2-devel or something. Then re-build the portaudio module and library (a couple well placed make cleans should do it). Andrew ___ 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] Sipura Codec Problem
I am willing to support this with the note that its incorrect and will not support it by default but update to trunk and try: param name=NDLB-allow-bad-iananame value=true/ this should fix it for you, SIGH On Tue, Nov 3, 2009 at 12:27 PM, Brian West br...@freeswitch.org wrote: Sounds like bad planning. I would send out a memo to your users and have them fix it. I have raised a bug multiple times with Cisco g729a is NOT valid. /b On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: Yes, that was my first option, but there many endpoints that I'm not able to configure. Basically it's a broadband solution where I have like 1000 endpoints that are out of my provisioning. Thanks, M ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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] Error checking for PMP [general error]
If you don't have a router with NAT-PMP enabled then it is expected. Same if you don't have upnp. If you are behind a NAT, it is in your best interest to enable one or the other in your router. It will save you a bunch of headaches... On Tue, Nov 3, 2009 at 3:25 PM, Jerry Richards jerry.richa...@teotech.com wrote: When I start Freeswitch, I see an Error checking for PMP [general error] as shown below. Does anyone know what could cause this? [r...@teoproxy bin]# ./freeswitch Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch -waste. auto-adjusting stack size for optimal performance 2009-11-02 10:12:27.17579 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-02 10:12:27.18373 [DEBUG] switch_event.c:553 Create event dispatch thread 0 2009-11-02 10:12:27.428749 [INFO] switch_nat.c:392 Scanning for NAT 2009-11-02 10:12:27.428885 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 2009-11-02 10:12:27.678480 [DEBUG] switch_nat.c:152 Checking for PMP 2/5 2009-11-02 10:12:27.679449 [DEBUG] switch_nat.c:152 Checking for PMP 3/5 2009-11-02 10:12:28.179388 [DEBUG] switch_nat.c:152 Checking for PMP 4/5 2009-11-02 10:12:29.179217 [DEBUG] switch_nat.c:152 Checking for PMP 5/5 2009-11-02 10:12:31.178879 [ERR] switch_nat.c:183 Error checking for PMP [general error] 2009-11-02 10:12:31.178902 [DEBUG] switch_nat.c:397 Checking for UPnP 2009-11-02 10:12:43.176881 [INFO] switch_nat.c:411 No PMP or UPnP NAT detected! 2009-11-02 10:12:43.210145 [INFO] switch_core_sqldb.c:538 Opening DB 2009-11-02 10:12:43.919804 [NOTICE] switch_scheduler.c:166 Starting task thread 2009-11-02 10:12:43.937881 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1257185563 2009-11-02 10:12:43.937980 [CONSOLE] switch_core.c:1449 Bringing up environment. 2009-11-02 10:12:43.937994 [CONSOLE] switch_core.c:1450 Loading Modules. 2009-11-02 10:12:43.938319 [INFO] switch_time.c:661 Timezone loaded 530 definitions 2009-11-02 10:12:43.938336 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_SOFTTIMER_MODULE] 2009-11-02 10:12:43.938351 [NOTICE] switch_loadable_module.c:228 Adding Timer 'soft' 2009-11-02 10:12:43.938413 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_PCM_MODULE] Best Regards, Jerry ___ 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 -- -Rupa ___ 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] IP Phones with FreeSwitch
Rupa, thanks a lot for the pointers - I'm just about to try to pick up some phones, so the tips are timely. Actually, I have been trying the IRC thing today, but keep getting connection refused, it's been a few years since I used IRC, but I think I have a Firewall problem that I'm working on. Hopefully, I'll be there soon, regards Dave - Original Message - From: Rupa Schomaker r...@rupa.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, November 03, 2009 9:19 PM Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch These phones work with FS, come by irc and you can talk to sekil about his use of them. In general, if you haven't invested in a bunch of phones, I'd recommend: Polycom 330,450,550 -- pick your price point snom - again, pick your price point These are generally well supported over the rest. On Tue, Nov 3, 2009 at 2:57 PM, Dave Stevenson steve...@primrosebank.net wrote: Hi again, sorry to be here again ! OK, now that I know that 3Com phones and FreeSwitch don't mix, my next question is about Cisco ! I see that the FreeSwitch Interoperability list includes Cisco phones such as the 7940 and 7960. I believe that these phones need user licenses to work with Cisco Call Manager. What I'd like to confirm is that I would not need any Cisco licenses or anything else to get a Cisco IP phone working with FreeSwitch. Again, I'd really appreciate feedback from anyone using either of these (or other) Cisco phones with FreeSwitch on whether any additional licenses or software are required to work with an out of the box FreeSwitch installation ? regards Dave ___ 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 -- -Rupa ___ 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 ___ 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] IP Phones with FreeSwitch
FYI, you can't do presence with the Cisco phones, so you can't see if someone is on the phone. -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Dave Stevenson Sent: Tuesday, November 03, 2009 3:29 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch Thanks a lot - to both William and Shelby, that makes me more confident about trying out at least one Cisco and Rupa has just given me a few more options, so, hopefully, I won't make the 3Com mistake again ! regards Dave - Original Message - From: William Suffill william.suff...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, November 03, 2009 9:10 PM Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch Cisco 7960 and the like that they push on the enterprise level for call manager also can be flashed with sip based firmware. I've only used the 7960 with the sip firmware. SPA942 and the like that used to be under Linksys/Sipura before that are targeted more toward smaller businesses and run SIP out of the box without any license complications. -- W ___ 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 ___ 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 ___ 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] Setting up Conference with Moderator
Was that sarcasm or you really mean it? -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian West Sent: Monday, November 02, 2009 9:08 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Setting up Conference with Moderator you know I have heard this before... It seems to ONLY be ATT /b On Nov 2, 2009, at 9:54 AM, Ujjval Karihaloo wrote: Yes, I think I did. However here is what furthur testing revelas. If I dial in from ATT cell phone, I do not see any DTMF using Don's IVR.xml.conf to call my conf app. But when I dial the same number using a Verizon Cell, it works. When I dial a number that is provisioned to call the Conf App directly from the public.xml dialplan...it works even with the same ATT cell phone... Strange behaviour ___ 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 ___ 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 REINVITE on Blind Transfer with bypass_media
Do you have ANY nat involved? /b On Nov 3, 2009, at 6:05 PM, Humberto Quintana wrote: 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/ external/514xxx...@a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] after sending the ACK for the reINVITE ___ 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 REINVITE on Blind Transfer with bypass_media
I don't know what you are talking about anymore. The scenario I had tested is when a call is bridged in bypass_media=true bridge and you blind transfer that call back to the dialplan as soon as it hits the routing state it will resume media. it has been confirmed to not work and confirmed to have been fixed several time and if you are still having a problem you must have something blocking some of your packets or something . You have to understand that sip is a protocol and your description is completely non-standard. Perhaps you should get a console trace and attach it to a jira. The trace probably makes more sense to me. sofia profile internal siptrace on console loglevel debug reproduce and attach the whole capture. On Tue, Nov 3, 2009 at 6:05 PM, Humberto Quintana hjqlo...@hotmail.comwrote: Hi, I tried r15332 and set in the sofia profile: a) bypass_media_after_bridge=true only b) bypass_media_after_bridge=true, param name=media-option value=resume-media-on-hold/ param name=media-option value=bypass-media-after-att-xfer/ In both cases FS is hanging up the initial call (A to FS) after accepting the REFER to C: A - reINVITE with FS' SDP - FS A - 200 - FS A - ACK - FS A - BYE - FS The call to C is not even tried. I found this line is the logs that could give some idea: 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/external/514xxx...@a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] after sending the ACK for the reINVITE Regards, Humberto please try r15326 I think i have it working. I recommend for optimal results you set bypass_media_after_bridge=true either as a global or in your DP in place of bypass_media=true On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana hjqlopez at hotmail.comwrote: Hi Mike, I re-tried with trunk rev 15319 but I got almost the same behavior: There is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But still there is no reINVITE for A (with C's SDP) after the call from FS to C is established. Anyway, we decided for now to do a different implementation but if you want to explore more in this issue count me in ;-) Thank you very much! Humberto _ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://go.microsoft.com/?linkid=9691817 ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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] Sipura Codec Problem
THE BLOODY MADNESS!!! I can only stop if people start saying 'NO'. :) /b On Nov 3, 2009, at 1:21 PM, Anthony Minessale wrote: Don't forget the one where there was a typo in the one for G722 so now we are all required to emulate that typo by running a 16khz codec with 8khz timestamps and sdp params. ___ 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] Sipura Codec Problem
At some point the paint will be rubbed off the magic lamp. /b On Nov 3, 2009, at 1:11 PM, Kristian Kielhofner wrote: It appears that Tony has already added an option (amazing) BUT you should really be setup for central provisioning with an installed base that large... You'll eventually have issues that *NO* amount of Tony/FreeSWITCH magic can fix. ___ 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] 3Com 3102 (3C10402B) Phone with FreeSwitch
Chris, thanks a lot for the response. It's not the answer that I wanted, but it is what I was coming round to thinking. As much as I'm disappointed (particularly as I've just got the phone), but at least it's a definitive answer and I can avoid wasting any more time with it, so thanks again. Oh well, off to try and find some open SIP phones that will actually work for me, regards Dave - Original Message - From: Chris Chen To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, November 03, 2009 8:18 PM Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch I think you are most likely on the wrong track, 3COM phones are locked to either 3COM PBX or the special Asterisk edition locked-down by 3COM. You cannot make them work with either FreeSWITCH or any other open SIP server other than 3COM IP PBX systems. I learned this over one year ago by playing with 3COm 3102 phones myself. Chris On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson steve...@primrosebank.net wrote: Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a special version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave - Original Message - From: Tihomir Culjaga To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, November 03, 2009 7:53 PM Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson steve...@primrosebank.net wrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave ___ 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 -- ___ 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 ___ 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 -- ___ 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 ___ 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] IP Phones with FreeSwitch
Cisco 7960 and the like that they push on the enterprise level for call manager also can be flashed with sip based firmware. I've only used the 7960 with the sip firmware. SPA942 and the like that used to be under Linksys/Sipura before that are targeted more toward smaller businesses and run SIP out of the box without any license complications. -- W ___ 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
[Freeswitch-users] Newbie trying to setup Cisco 7940 phones
I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin dsi sh conf -- Current *FLASH* Configuration -- Platform : Cisco Systems, Inc. IP Phone CP-7940G Elapsed Time: 01:01:06 dhcp_server : Disabled my_ip_addr : 10.86.11.50 subnet_mask : 255.255.0.0 defaultgw : 10.86.0.1 dyn_dns_addr_1 : 0.0.0.0 dyn_dns_addr_2 : 0.0.0.0 dns_addr : 10.85.0.11 dns_backup_1: 10.85.0.10 primary_tftp_addr : 10.86.10.58 dyn_tftp_addr : 0.0.0.0 my_mac_addr : 0012:7f98:eaa9 domain_name : dsi-corp.net my_name : SIP00127F98EAA9 Status Flags : 1231 image_version : P003-8-12-00 FirmLoadID : PC030301 DSPLoadID : PS03AT38 network_media_type : Auto network_port2_type : Hub/Switch dscpForAudio : 184 phone_label : Matthew Kitchin tftp_cfg_dir : phone_password : ** phone_prompt : dsi language : english sntp_mode : Unicast sntp_server : 10.85.0.10 time_zone : CST dst_offset : 01/00 dst_start_month : March dst_start_day : 0 dst_start_day_of_week : Sunday dst_start_week_of_month : 8 dst_start_time : 02/00 dst_stop_month : October dst_stop_day : 0 dst_stop_day_of_week : Sunday dst_stop_week_of_month : 8 dst_stop_time : 02/00 dst_auto_adjust : 1 time_format_24hr : 1 date_format : M/D/Y nat_enable : 0 nat_address : UNPROVISIONED voip_control_port : 5060 start_media_port : 16384 end_media_port : 32766 sync : 1 xml_card_dir : xml_card_file : CARD.XML telnet_level : 1 services_url : directory_url : logo_url : http_proxy_addr : UNPROVISIONED http_proxy_port : 80 garp_enable : 0 enable_vad : 0 dial_template : dialplan callerid_blocking : 0 anonymous_call_block : 0 autocomplete : 0 messages_uri : dnd_control : 2 preferred_codec : g711ulaw dtmf_outofband : avt_always dtmf_avt_payload : 101 dtmf_db_level : 3 dtmf_inband : 0 call_manager1_addr : UNPROVISIONED call_manager2_addr : UNPROVISIONED call_manager3_addr : UNPROVISIONED call_manager1_sip_port : 5060 call_manager2_sip_port : 5060 call_manager3_sip_port : 5060 call_manager5_addr : UNPROVISIONED call_manager5_sip_port : 5060 call_manager4_addr : UNPROVISIONED call_manager4_sip_port : 0 line1_name : 1008 line2_name : 1001 line1_authname : 1008 line2_authname : 1001 line1_password : ** line2_password : ** line1_shortname : 1008 line2_shortname : 1001 line1_displayname : 1008 line2_displayname : UNPROVISIONED line1_contact : UNPROVISIONED line2_contact : UNPROVISIONED proxy1_address : nshplpbx1.unix proxy2_address : nshplpbx1.unix proxy1_port : 5060 proxy2_port : 5060 sip_retx : 10 sip_invite_retx : 6 timer_t1 : 500 timer_t2 : 4000 timer_invite_expires : 180 timer_register_expires : 3600 proxy_register : 1 proxy_backup : proxy_emergency : proxy_backup_port : 5060 proxy_emergency_port : 5060 outbound_proxy : UNPROVISIONED outbound_proxy_port : 5060 nat_received_processing : 0 mwi_status : 0 call_waiting : 1 user_info : phone cnf_join_enable : 0 remote_party_id : 1 semi_attended_transfer : 1 transfer_onhook_enabled : 0 call_hold_ringback : 3 stutter_msg_waiting : 0 cfwd_url : call_stats : 0 auto_answer : 0 local_cfwd_enable : 1 timer_register_delta : 5 sip_max_forwards : 70 rfc_2543_hold : 0 version_stamp : timer_keepalive_expires : 120 connection_monitor_duration : 120 encrypt_key : ** 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5067 0 acls to check for proxy 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5085 network ip is a proxy [0] 2009-11-03 15:39:50.762957 [DEBUG] sofia.c:5113 IP 10.86.10.58 Rejected by acl domains. Falling back to Digest auth. 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5067 0 acls to check for proxy 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5085 network ip is a proxy [0] 2009-11-03 15:39:50.780028 [DEBUG] sofia.c:5113 IP 10.86.10.58 Rejected by acl domains. Falling back to Digest auth. 2009-11-03 15:39:50.797901 [NOTICE] switch_channel.c:613 New Channel sofia/internal/1...@nshplpbx1.unix [8c133ad4-67cf-4ffa-8655-56ffa0e3933d] 2009-11-03
Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch
Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a special version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave - Original Message - From: Tihomir Culjaga To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, November 03, 2009 7:53 PM Subject: Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson steve...@primrosebank.net wrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave ___ 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 -- ___ 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 ___ 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 REINVITE on Blind Transfer with bypass_media
Hi, I tried r15332 and set in the sofia profile: a) bypass_media_after_bridge=true only b) bypass_media_after_bridge=true, param name=media-option value=resume-media-on-hold/ param name=media-option value=bypass-media-after-att-xfer/ In both cases FS is hanging up the initial call (A to FS) after accepting the REFER to C: A - reINVITE with FS' SDP - FS A - 200 - FS A - ACK - FS A - BYE - FS The call to C is not even tried. I found this line is the logs that could give some idea: 2009-11-03 18:29:41.280707 [NOTICE] mod_sofia.c:733 Hangup sofia/external/514xxx...@a.b.c.d [CS_ROUTING] [RECOVERY_ON_TIMER_EXPIRE] after sending the ACK for the reINVITE Regards, Humberto please try r15326 I think i have it working. I recommend for optimal results you set bypass_media_after_bridge=true either as a global or in your DP in place of bypass_media=true On Mon, Nov 2, 2009 at 4:30 PM, Humberto Quintana hjqlopez at hotmail.comwrote: Hi Mike, I re-tried with trunk rev 15319 but I got almost the same behavior: There is now a reINVITE (with FS' SDP) going to A when the REFER is accepted. But still there is no reINVITE for A (with C's SDP) after the call from FS to C is established. Anyway, we decided for now to do a different implementation but if you want to explore more in this issue count me in ;-) Thank you very much! Humberto _ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://go.microsoft.com/?linkid=9691817 ___ 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] IP Phones with FreeSwitch
Any of the Cisco phones with a SIP image should work fine ... no license required. SDR Dave Stevenson wrote: Hi again, sorry to be here again ! OK, now that I know that 3Com phones and FreeSwitch don't mix, my next question is about Cisco ! I see that the FreeSwitch Interoperability list includes Cisco phones such as the 7940 and 7960. I believe that these phones need user licenses to work with Cisco Call Manager. What I'd like to confirm is that I would not need any Cisco licenses or anything else to get a Cisco IP phone working with FreeSwitch. Again, I'd really appreciate feedback from anyone using either of these (or other) Cisco phones with FreeSwitch on whether any additional licenses or software are required to work with an out of the box FreeSwitch installation ? regards Dave ___ 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 ___ 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] mod_valet_parking: auto reports on wrong leg of call
There are 2 ways to use the auto in one is to attended transfer the call into the extension with auto in the other is to bind_meta_app a call to valet_park + auto in blind transfer to auto in only has one leg so the guy you transferred is the only one who can hear it because when you press the blind xfer key you hangup the call on your side. On Tue, Nov 3, 2009 at 3:28 AM, Brian Stafford brian.staff...@lattice-voice.com wrote: Brian Stafford wrote: Brian West wrote: You have to be doing it wrong then. Can you show us your dialplan you should have two extensions one for the lot range and one to attended transfer someone into the lot. /b The relevant excerpt from the dialplan is extension name=valet_unpark condition field=destination_number expression=^(41[0-9])$ action application=answer/ action application=valet_park data=valet_lot $1/ /condition /extension extension name=valet_park condition field=destination_number expression=^(420)$ action application=answer/ action application=valet_park data=valet_lot auto in 410 419/ /condition /extension x410-419 are the slots and 420 parks a call. Parking by picking one of 410-419 works fine and subsequently dialling them from another works fine, I added x420 for the auto feature. Regards Brian _ Any clues what I'm doing wrong? Is more information needed? Brian ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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
[Freeswitch-users] Error checking for PMP [general error]
When I start Freeswitch, I see an Error checking for PMP [general error] as shown below. Does anyone know what could cause this? [r...@teoproxy bin]# ./freeswitch Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch -waste. auto-adjusting stack size for optimal performance 2009-11-02 10:12:27.17579 [INFO] switch_event.c:565 Activate Eventing Engine. 2009-11-02 10:12:27.18373 [DEBUG] switch_event.c:553 Create event dispatch thread 0 2009-11-02 10:12:27.428749 [INFO] switch_nat.c:392 Scanning for NAT 2009-11-02 10:12:27.428885 [DEBUG] switch_nat.c:152 Checking for PMP 1/5 2009-11-02 10:12:27.678480 [DEBUG] switch_nat.c:152 Checking for PMP 2/5 2009-11-02 10:12:27.679449 [DEBUG] switch_nat.c:152 Checking for PMP 3/5 2009-11-02 10:12:28.179388 [DEBUG] switch_nat.c:152 Checking for PMP 4/5 2009-11-02 10:12:29.179217 [DEBUG] switch_nat.c:152 Checking for PMP 5/5 2009-11-02 10:12:31.178879 [ERR] switch_nat.c:183 Error checking for PMP [general error] 2009-11-02 10:12:31.178902 [DEBUG] switch_nat.c:397 Checking for UPnP 2009-11-02 10:12:43.176881 [INFO] switch_nat.c:411 No PMP or UPnP NAT detected! 2009-11-02 10:12:43.210145 [INFO] switch_core_sqldb.c:538 Opening DB 2009-11-02 10:12:43.919804 [NOTICE] switch_scheduler.c:166 Starting task thread 2009-11-02 10:12:43.937881 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1257185563 2009-11-02 10:12:43.937980 [CONSOLE] switch_core.c:1449 Bringing up environment. 2009-11-02 10:12:43.937994 [CONSOLE] switch_core.c:1450 Loading Modules. 2009-11-02 10:12:43.938319 [INFO] switch_time.c:661 Timezone loaded 530 definitions 2009-11-02 10:12:43.938336 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_SOFTTIMER_MODULE] 2009-11-02 10:12:43.938351 [NOTICE] switch_loadable_module.c:228 Adding Timer 'soft' 2009-11-02 10:12:43.938413 [CONSOLE] switch_loadable_module.c:889 Successfully Loaded [CORE_PCM_MODULE] Best Regards, Jerry ___ 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] 3Com 3102 (3C10402B) Phone with FreeSwitch
I think you are most likely on the wrong track, 3COM phones are locked to either 3COM PBX or the special Asterisk edition locked-down by 3COM. You cannot make them work with either FreeSWITCH or any other open SIP server other than 3COM IP PBX systems. I learned this over one year ago by playing with 3COm 3102 phones myself. Chris On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson steve...@primrosebank.netwrote: Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a special version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave - Original Message - *From:* Tihomir Culjaga tculj...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, November 03, 2009 7:53 PM *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson steve...@primrosebank.netwrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave ___ 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 -- ___ 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 ___ 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 ___ 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] 3Com 3102 (3C10402B) Phone with FreeSwitch
well, if it is a sip phone than you should be able to input your usernamepassword somewhere. Usually, SIP phones downloads their configuration using dhcp/tftp|http method... the FW is downloaded just once if you need to upgrade the phone... I don't have any of these phones on my desk, just found the manual on the web. anyhow, freeswitch is expecting a SIP phone to register and thats it :P ... there is no specific phone provisioning from FS side. T. On Tue, Nov 3, 2009 at 9:03 PM, Dave Stevenson steve...@primrosebank.netwrote: Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a special version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave - Original Message - *From:* Tihomir Culjaga tculj...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, November 03, 2009 7:53 PM *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson steve...@primrosebank.netwrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave ___ 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 -- ___ 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 ___ 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 ___ 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] IP Phones with FreeSwitch
Thanks a lot - to both William and Shelby, that makes me more confident about trying out at least one Cisco and Rupa has just given me a few more options, so, hopefully, I won't make the 3Com mistake again ! regards Dave - Original Message - From: William Suffill william.suff...@gmail.com To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, November 03, 2009 9:10 PM Subject: Re: [Freeswitch-users] IP Phones with FreeSwitch Cisco 7960 and the like that they push on the enterprise level for call manager also can be flashed with sip based firmware. I've only used the 7960 with the sip firmware. SPA942 and the like that used to be under Linksys/Sipura before that are targeted more toward smaller businesses and run SIP out of the box without any license complications. -- W ___ 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 ___ 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] WARNING On Inbound Call Question
can you try the same thing with the latest trunk or pre-release tarball.
On Tue, Nov 3, 2009 at 3:35 PM, Jerry Richards
jerry.richa...@teotech.comwrote:
I have my Freeswitch server with an installed Sangoma A101D card. Most
everything works okay, however, when I get an inbound call from the PSTN, I
see the following warning show up in the log. Additionally, the caller (on
the PSTN) does not hear ringback, and if the call is not answered within
about 12 seconds, the call ends (so it doesn't go to voice mail). If I
make
a call from one internal phone to another, then it will go to voice mail
after 30 seconds.
Here are the two warnings:
[WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1]
Rc=0 CSid=0 Seq=11
[WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to
PROGRESS_MEDIA
Here is the log of the warning upon an inbound call:
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com 2009-11-02 09:06:01.664835
[WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0
Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176]
2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on
1:1 from DOWN to RING
2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig
[START]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound
channel OpenZAP/1:1/5384
2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel
OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384)
State Change CS_NEW - CS_INIT
2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal
OpenZAP/1:1/5384 [BREAK]
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398
(OpenZAP/1:1/5384) Running State Change CS_INIT
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481
(OpenZAP/1:1/5384) State INIT
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384)
State Change CS_INIT - CS_ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal
OpenZAP/1:1/5384 [BREAK]
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481
(OpenZAP/1:1/5384) State INIT going to sleep
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398
(OpenZAP/1:1/5384) Running State Change CS_ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484
(OpenZAP/1:1/5384) State ROUTING
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384
CHANNEL ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78
OpenZAP/1:1/5384 Standard ROUTING
2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing
4253813176-5384 in context default
Dialplan: OpenZAP/1:1/5384 parsing [default-unloop] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~
/^true$/ break=on-false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~
/^true$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-tod_example] continue=true
Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example]
Dialplan: OpenZAP/1:1/5384 Action set(open=true)
Dialplan: OpenZAP/1:1/5384 parsing [default-SangomaPRI] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI]
destination_number(5384) =~ /^9(\d+)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-global-intercept]
continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept]
destination_number(5384) =~ /^(5380)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-group-intercept]
continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept]
destination_number(5384) =~ /^\*8$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-intercept-ext] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext]
destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-redial] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384)
=~
/^870$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-global] continue=true
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~
/^true$/ break=never
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: OpenZAP/1:1/5384 Absolute Condition [global]
Dialplan: OpenZAP/1:1/5384 Action
[Freeswitch-users] Dial Plan Question
My understanding of DialPlan/CallRouting is that it can be accomplished via static XML tags, or alternatively, via a DialPlan Application that interfaces with the dptools module. Question: If my above assumption is true, how does one select one approach over the other? What is the criteria/considerations that would govern the decision? Best Regards, Jerry ___ 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] 3Com 3102 (3C10402B) Phone with FreeSwitch
pity,the phone looks quite nice... On Tue, Nov 3, 2009 at 9:18 PM, Chris Chen chris.chen2...@gmail.com wrote: I think you are most likely on the wrong track, 3COM phones are locked to either 3COM PBX or the special Asterisk edition locked-down by 3COM. You cannot make them work with either FreeSWITCH or any other open SIP server other than 3COM IP PBX systems. I learned this over one year ago by playing with 3COm 3102 phones myself. Chris On Tue, Nov 3, 2009 at 3:03 PM, Dave Stevenson steve...@primrosebank.netwrote: Tihomir, thanks for the link, but actually, I had already found/downloaded/read and almost understood that document ! However, the options to log into the phone and configure the extension number etc. do not appear on my phone. From reading another post on the web, I don't think that the phone has the SIP software loaded until it is downloaded from the Server - I think that there is a special version of Asterix for 3Com that does this, maybe the same functionality does not exist in FreeSwitch ? Maybe I should have been clearer in the post below, but I think that this is the root of the problem. I think that the 3Com phone is looking for the Switch to download the SIP firmware to it and FreeSwitch does not seem to do that. Given that you have pointed me in the direction of that document, are you using 3Com Phones with FreeSwitch ? If so, I'm obviously on the wrong track, but please let me know how you've made it work regards Dave - Original Message - *From:* Tihomir Culjaga tculj...@gmail.com *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tuesday, November 03, 2009 7:53 PM *Subject:* Re: [Freeswitch-users] 3Com 3102 (3C10402B) Phone with FreeSwitch you might read this before you bigin :P http://support.3com.com/documents/asterisk/Asterisk_TeleGd_Business_AB.pdf T. On Tue, Nov 3, 2009 at 6:30 PM, Dave Stevenson steve...@primrosebank.net wrote: Help please . . . . Is anyone using the 3Com 3102 (3C10402B) Phone with FreeSwitch ? I have got FreeSwitch up and running with the SoftPhone, but can't get a 3Com hardware phone to talk to FreeSwitch. I have the phone getting its IP Address from DHCP and it can see the FreeSwitch server but I can't find anything in the phone to allow the extension password to be configured. Can FreeSwitch send this data to the phone (and if so, which configuration files are involved) or must the phone be configured manually before it can talk to FreeSwitch ? Any help would be really appreciated as I'm pulling my hair out here ! Regards Dave ___ 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 -- ___ 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 ___ 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 ___ 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 ___ 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] Sipura Codec Problem
Don't forget the one where there was a typo in the one for G722 so now we are all required to emulate that typo by running a 16khz codec with 8khz timestamps and sdp params. On Tue, Nov 3, 2009 at 1:12 PM, Arsen Chaloyan achalo...@yahoo.com wrote: Actually, there were a few more misinterpretations in earlier software of Cisco Gateways, which RFC implementers had to address in RFC3551, strange ... RTP Payload Type 19 remains reserved because some implementations wrongly interpreted 13 decimal as 13 hexadecimal value. Another issue is G726 bit packing. Again some implementations used wrong bit packing and RFC3551 tried to partially resolve this conflict introducing new payload format named AAL2-G726 ... -- *From:* Brian West br...@freeswitch.org *To:* freeswitch-users@lists.freeswitch.org *Sent:* Tue, November 3, 2009 10:27:39 PM *Subject:* Re: [Freeswitch-users] Sipura Codec Problem Sounds like bad planning. I would send out a memo to your users and have them fix it. I have raised a bug multiple times with Cisco g729a is NOT valid. /b On Nov 3, 2009, at 12:11 PM, Mariano de Llano wrote: Yes, that was my first option, but there many endpoints that I'm not able to configure. Basically it's a broadband solution where I have like 1000 endpoints that are out of my provisioning. Thanks, M ___ 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 ___ 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 -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ 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] IP Phones with FreeSwitch
These phones work with FS, come by irc and you can talk to sekil about his use of them. In general, if you haven't invested in a bunch of phones, I'd recommend: Polycom 330,450,550 -- pick your price point snom - again, pick your price point These are generally well supported over the rest. On Tue, Nov 3, 2009 at 2:57 PM, Dave Stevenson steve...@primrosebank.net wrote: Hi again, sorry to be here again ! OK, now that I know that 3Com phones and FreeSwitch don't mix, my next question is about Cisco ! I see that the FreeSwitch Interoperability list includes Cisco phones such as the 7940 and 7960. I believe that these phones need user licenses to work with Cisco Call Manager. What I'd like to confirm is that I would not need any Cisco licenses or anything else to get a Cisco IP phone working with FreeSwitch. Again, I'd really appreciate feedback from anyone using either of these (or other) Cisco phones with FreeSwitch on whether any additional licenses or software are required to work with an out of the box FreeSwitch installation ? regards Dave ___ 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 -- -Rupa ___ 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] Precompiled Windows Binaries
Dave, Carlos can probably be a better help here too but yes Freepbx v3 is a web gui that is under heavy development - it probably is not yet ready for production but looks very promising! you can navigate to http://127.0.0.1/freepbx-v3/index.php/installer.html and restart the installer for freepbx if you want to experiment with it. The base FreeSWITCH installer does install and work well with windows and is quite easy to learn and configure. Their is a lot to learn though :) Regards, Jeff Dave Stevenson wrote: Jeff, thanks a lot for the reply. I was a little confused by the fact that the SVN Snapshot was some 10MB smaller than the Full 1.0.4 file so worried that I might lose something. As you say though, think that I'll cross my fingers and try the updated release. I am running FreeSwitch on a test machine at the moment until the target hardware arrives - hopefully tomorrow, so I can afford to have a little play. You mentioned FreePBX V3. I had been fumbling around trying to work out what this is and from what I've read, it seems to provide a GUI Front End for configuring FreeSwitch ? I am guessing that while it has been installed with FreeSwitch, I then need to run the FreePBX Installer to update the FreePBX/FreeSwitch configuration on my hardware ? When I start FreeSwitch, it does not automatically load the WAMPServer. When I start WAMPServer manually, and open up localhost (127.0.0.1) in a web browser, I can see the WampServer logo and various tools such as phpinfo() and phpmyadmin. FreePBX is there under Your Projects. When I opened this up the first time, it appeared to want to install FreePBX over FreeSwitch, I tried to abort this when it was going to overwrite some FreeSwitch conf files and I thought I'd better not go on until I had a better idea what was happening. I backed out of the FreePBX install and now I can't get the FreePBX or phpmyadmin pages up again (missing files) so it looks like I'm going to have to reinstall anyway. So, for next time,am I right in thinking that I should proceed with running the FreePBX install from the WAMPServer menu ? regards Dave - Original Message - From: Jeff Lenk jl...@frontiernet.net To: freeswitch-users@lists.freeswitch.org Sent: Tuesday, November 03, 2009 2:48 PM Subject: Re: [Freeswitch-users] Precompiled Windows Binaries Hi Dave, These are supported by Carlos Talbot . They also include Freepbx v3 Just as you said freeswitch-1.0.4.exe is the tagged release and freeswitch.exe is a newer svn snapshot. There should be no problems installing the new version allthough best to just try and see! Not sure why the newest one is from October 7th. Jeff Dave Stevenson wrote: Hi, I have read the Docs on the Wiki (http://wiki.freeswitch.org/wiki/Installation_Guide#Precompiled_Binaries) but am still not sure of what the different Windows install files are. Currently, the Windows Installer directory contains :- LATEST_SVN_15106 - 6 Bytes freeswitch-1.0.4.exe - 42 Megabytes freeswitch.exe - 32 Megabytes I have installed the freeswitch-1.0.4.exe file which is dated 3rd September. The freeswitch.exe file is dated 7th October and think that it contains the minor updates since 3rd September ? Could someone who knows FreeSwitch under windows help me understand the two files please ? I chickened out of running the later exe in case it did something to the running install of FreeSwitch 1.0.4, is it safe to run the newer exe with the old one already installed ? What will it actually do ? regards Dave ___ 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 -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3938887.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ 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 ___ 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 -- View this message in context: http://n2.nabble.com/Precompiled-Windows-Binaries-tp3937943p3940700.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org
Re: [Freeswitch-users] Sipura Codec Problem
It appears that Tony has already added an option (amazing) BUT you should really be setup for central provisioning with an installed base that large... You'll eventually have issues that *NO* amount of Tony/FreeSWITCH magic can fix. On Tue, Nov 3, 2009 at 1:11 PM, Mariano de Llano mariano.dell...@gmail.com wrote: Yes, that was my first option, but there many endpoints that I'm not able to configure. Basically it's a broadband solution where I have like 1000 endpoints that are out of my provisioning. Thanks, M -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com ___ 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
[Freeswitch-users] FS and Skinny (SCCP)
FS doesnt support SCCP (from what I gathered, just because no one has bothered coding it). Are there other users out there has use SCCP and FS? (with some middleware in between) If enough people would find a use for it, I'd be willing to actually code it (esp if someone offered a bounty). So, would anyone besides me want/use a SCCP endpoint in FS? -- mm_202. ___ 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] Sipura Codec Problem
Yah this one is LLLAME :P We have some dyslexic engineers. /b On Nov 3, 2009, at 1:12 PM, Arsen Chaloyan wrote: Another issue is G726 bit packing. Again some implementations used wrong bit packing and RFC3551 tried to partially resolve this conflict introducing new payload format named AAL2-G726 ... ___ 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
[Freeswitch-users] IP Phones with FreeSwitch
Hi again, sorry to be here again ! OK, now that I know that 3Com phones and FreeSwitch don't mix, my next question is about Cisco ! I see that the FreeSwitch Interoperability list includes Cisco phones such as the 7940 and 7960. I believe that these phones need user licenses to work with Cisco Call Manager. What I'd like to confirm is that I would not need any Cisco licenses or anything else to get a Cisco IP phone working with FreeSwitch. Again, I'd really appreciate feedback from anyone using either of these (or other) Cisco phones with FreeSwitch on whether any additional licenses or software are required to work with an out of the box FreeSwitch installation ? regards Dave___ 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] Dial Plan Question
I think the real question is what are you trying to do ... for some things it's very easy to just whip up a static XML file and be done with it. For others you probably want some sort of interaction with a DB. The options here are pretty endless: -- XML curl -- handing off the call to a script call from a static dial plan (use lua if there is going to be any load) -- event_socket -- mod_lcr But ultimately I think it's what you're trying to accomplish that matters. For a PBX install I'd say static files is probably about as easy as it is going to get. For delivering a service you'd probably want interaction with a DB. I've use XML curl a lot and have even starting using direct DB queries from static dialplans using mod_memcache and memcachedb (not memcache ... persistent storage). SDR Jerry Richards wrote: My understanding of DialPlan/CallRouting is that it can be accomplished via static XML tags, or alternatively, via a DialPlan Application that interfaces with the dptools module. Question: If my above assumption is true, how does one select one approach over the other? What is the criteria/considerations that would govern the decision? Best Regards, Jerry ___ 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 ___ 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
[Freeswitch-users] WARNING On Inbound Call Question
I have my Freeswitch server with an installed Sangoma A101D card. Most
everything works okay, however, when I get an inbound call from the PSTN, I
see the following warning show up in the log. Additionally, the caller (on
the PSTN) does not hear ringback, and if the call is not answered within
about 12 seconds, the call ends (so it doesn't go to voice mail). If I make
a call from one internal phone to another, then it will go to voice mail
after 30 seconds.
Here are the two warnings:
[WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_START_ACK:(81) [w1g1]
Rc=0 CSid=0 Seq=11
[WARNING] mod_openzap.c:761 VETO Changing state on 1:1 from PROGRESS to
PROGRESS_MEDIA
Here is the log of the warning upon an inbound call:
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com
freeswi...@teoproxy.greyhawk.tonecommander.com 2009-11-02 09:06:01.664835
[WARNING] ozmod_ss7_boost.c:1141 RX EVENT: CALL_START:(80) [w1g1] CSid=0
Seq=12 Cn=[N/A] Cd=[5384] Ci=[4253813176]
2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:655 Changing state on
1:1 from DOWN to RING
2009-11-02 09:06:01.665824 [DEBUG] ozmod_ss7_boost.c:841 1:1 STATE [RING]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1481 got clear channel sig
[START]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:344 Set codec PCMU 20ms
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1184 Connect inbound
channel OpenZAP/1:1/5384
2009-11-02 09:06:01.665824 [NOTICE] switch_channel.c:602 New Channel
OpenZAP/1:1/5384 [b678f311-ab74-4cc1-afac-b83d89a53132]
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:1192 (OpenZAP/1:1/5384)
State Change CS_NEW - CS_INIT
2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal
OpenZAP/1:1/5384 [BREAK]
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398
(OpenZAP/1:1/5384) Running State Change CS_INIT
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481
(OpenZAP/1:1/5384) State INIT
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:368 (OpenZAP/1:1/5384)
State Change CS_INIT - CS_ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_session.c:932 Send signal
OpenZAP/1:1/5384 [BREAK]
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:481
(OpenZAP/1:1/5384) State INIT going to sleep
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:398
(OpenZAP/1:1/5384) Running State Change CS_ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:484
(OpenZAP/1:1/5384) State ROUTING
2009-11-02 09:06:01.665824 [DEBUG] mod_openzap.c:391 OpenZAP/1:1/5384
CHANNEL ROUTING
2009-11-02 09:06:01.665824 [DEBUG] switch_core_state_machine.c:78
OpenZAP/1:1/5384 Standard ROUTING
2009-11-02 09:06:01.665824 [INFO] mod_dialplan_xml.c:315 Processing
4253813176-5384 in context default
Dialplan: OpenZAP/1:1/5384 parsing [default-unloop] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (PASS) [unloop] ${unroll_loops}(true) =~
/^true$/ break=on-false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [unloop] ${sip_looped_call}() =~
/^true$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-tod_example] continue=true
Dialplan: OpenZAP/1:1/5384 Absolute Condition [tod_example]
Dialplan: OpenZAP/1:1/5384 Action set(open=true)
Dialplan: OpenZAP/1:1/5384 parsing [default-SangomaPRI] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [SangomaPRI]
destination_number(5384) =~ /^9(\d+)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-global-intercept]
continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global-intercept]
destination_number(5384) =~ /^(5380)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-group-intercept] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [group-intercept]
destination_number(5384) =~ /^\*8$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-intercept-ext] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [intercept-ext]
destination_number(5384) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-redial] continue=false
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [redial] destination_number(5384) =~
/^870$/ break=on-false
Dialplan: OpenZAP/1:1/5384 parsing [default-global] continue=true
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${call_debug}(false) =~
/^true$/ break=never
Dialplan: OpenZAP/1:1/5384 Regex (FAIL) [global] ${sip_has_crypto}() =~
/^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: OpenZAP/1:1/5384 Absolute Condition [global]
Dialplan: OpenZAP/1:1/5384 Action
hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid})
Dialplan: OpenZAP/1:1/5384 Action
hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe
r})
Dialplan: OpenZAP/1:1/5384 Action
hash(insert/${domain_name}-last_dial/global/${uuid})
Re: [Freeswitch-users] Sipura Codec Problem
On Tue, Nov 3, 2009 at 11:11 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: It appears that Tony has already added an option (amazing) BUT you should really be setup for central provisioning with an installed base that large... You'll eventually have issues that *NO* amount of Tony/FreeSWITCH magic can fix. Kristian is correct. Listen to him because he's familiar with having lots and lots of units out in the field. The bandage Tony applied will eventually wear off. The long-term solution is to treat the malady and not the symptom. I'm certain that members of the FS community could point you toward some resources to assist with central provisioning. -MC ___ 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
[Freeswitch-users] Wiki typo
Hi! Was just reading wiki here: http://wiki.freeswitch.org/wiki/Home_PBX_Example It lists sample sofia.conf.xml which has this parameter: !--param name=inbound-no-media value=true/-- I think it should read inbound-*bypass*-media and not inbound-*no*-media... I know, it says outdated but still, can be confusing. Anyone here who can edit wiki and correct? Thanks, Dmitry -- DG NJ ___ 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] Wiki typo
Yes you can login and edit the wiki yourself. Thanks, /b On Nov 3, 2009, at 9:16 PM, Dmitry Gromov wrote: Hi! Was just reading wiki here: http://wiki.freeswitch.org/wiki/Home_PBX_Example It lists sample sofia.conf.xml which has this parameter: !--param name=inbound-no-media value=true/-- I think it should read inbound-bypass-media and not inbound-no- media... I know, it says outdated but still, can be confusing. Anyone here who can edit wiki and correct? Thanks, Dmitry -- DG NJ ___ 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 ___ 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] Wiki typo
Thanks, done - page has been corrected! On Tue, Nov 3, 2009 at 22:34, Brian West br...@freeswitch.org wrote: Yes you can login and edit the wiki yourself. You know... I actually spent some time looking for login/create account link when I noticed this typo. No idea why I did not see it then :) -- DG NJ ___ 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] Newbie trying to setup Cisco 7940 phones
Michael Collins wrote: On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.pub...@gmail.com mailto:mkitchin.pub...@gmail.com mkitchin.pub...@gmail.com mailto:mkitchin.pub...@gmail.com wrote: I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53 http://10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now. Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd say first thing to do is capture the SIP traffic to see if there are any clues. A normal temporary failure doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org) You can also join the IRC channel #freeswitch on irc.freenode.net http://irc.freenode.net and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers. -MC Thank you. I think I did what you are looking for. I stopped FS and launched this command. TPORT_LOG=1 /usr/local/freeswitch/bin/freeswitch and captured all output to http://pastebin.freeswitch.org/10965 Does this tell you anything? I'm definitely new to SIP and phone system admin in general. I have plenty of network and Linux experience. With that in mind, someone on this mailing list emailed me directly and said SipX would be a better fit for me. Is that blasphemy for me to even mention? I went through the documentation and the provisioning aspect and web interface do look tempting to a novice. I apologize if this is like trying to buy a chevy at a ford dealership. I'm looking to deploy about 150 handsets at a corporate office and then 10 to 12 handsets at 120 remote locations. We are moving from an old key system, so our current features are very limited. We just need a few ACD groups, call history, and the other general basics. I first found Asterisk and read about some of the shortcomings. FS looks like the most robust solution. I have no idea where SipX would fit in. The people here are obviously a very knowledgeable group and I would gladly accept any thoughts, comments, etc. ___ 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] Wiki typo
On Tue, Nov 3, 2009 at 7:45 PM, Dmitry Gromov grom...@gmail.com wrote: Thanks, done - page has been corrected! On Tue, Nov 3, 2009 at 22:34, Brian West br...@freeswitch.org wrote: Yes you can login and edit the wiki yourself. You know... I actually spent some time looking for login/create account link when I noticed this typo. No idea why I did not see it then :) Thank you for not giving up! :) We appreciate it when the community helps out. Nicely done. -MC ___ 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] Newbie trying to setup Cisco 7940 phones
Matthew, I'm about in the same boat as you are, just on a smaller scale. We have a ton of Nortel telephony gear, but it's time to move out of the 90's and enter this millennium. My Cisco quote was in the same ballpark as yours. The Cisco stuff is mature, rock solid, meshes very well with their network gear and is actually relatively easy to set up and maintain if you know your way around IOS. I just refuse to pay that kind of money for yet another semi-proprietary solution. After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The developers and the community are great and available, but just starting out with SIP and voip in general, this may not be the best platform. So let the blasphemy begin :) SipX was a breeze to install (insert CD, boot, next next next...) and looks pretty solid. I believe they actually use FreeSWITCH for their voicemail and conferencing, internally. I just couldn't get my head around their GUI, ACD was too basic and had all kinds of issues getting stuff to just work. 3CX (Windows Only) was completely painless. It just worked. But I'm still not convinced that I want to run all my voice on a single windows box. Plus it's not free/open/etc and I don't want to lock myself in again. Although it's an asterisk based solution, I found trixbox to be very easy. Setup is automatic and everything just worked. The GUI is simple and logical enough that I can let somebody else handle the day-to-day phone setup and basic admin. I have my doubts about it scaling to 250 users, though. This may be a completely flawed strategy and I may very well be shooting myself in the foot by doing this, but I plan on piloting a trixbox install with a dozen or so users and see how stable it is. I'll keep a FreeSWITCH box next to it for the more advanced stuff. Once I get more comfortable with the intricacies of SIP and get some time to code a basic GUI for FreeSWITCH, I have a feeling that that trixbox is going to get phased out... Peter -Original Message- From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of mkitchin.pub...@gmail.com Sent: Tuesday, November 03, 2009 11:10 PM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones Michael Collins wrote: On Tue, Nov 3, 2009 at 2:19 PM, mkitchin.pub...@gmail.com mailto:mkitchin.pub...@gmail.com mkitchin.pub...@gmail.com mailto:mkitchin.pub...@gmail.com wrote: I'm working on an alternative to a $120,000 Cisco phone system that my company is looking at. I got Freeswitch installed on CentOS last week using the Quick and Dirty instructions. That part was painless. We had a few 7940s laying around. After some wrestling with it, I got the latest SIP firmware installed and what I hoped was a functional config (attached). X-Lite phones can call each other no problem. 7940s can call X-Lite no problem. Anytime I try and call a 7940, it goes straight to voicemail. I attached a log file that shows the activity when trying to call a7940 from X-Lite. X-Lite is at 10.86.10.58. 7940 is at 10.86.11.50. Freeswitch is nshplpbx1.unix/10.85.0.53 http://10.85.0.53. Everything is on the same LAN. Different subnets, but no firewalls. I didn't see anything that said posting attachments was frowned upon. I apologize if it isn't appropriate. I'm guessing this is something simple and I'm just clueless on how to diagnose the issue. I'm not tied to using this model for good, but it is what we had laying around. Any help would be greatly appreciated. Next step is configuring it to talk to Verizon VOIP over a DS3. Thanks, Matthew Kitchin Matthew, Welcome to FreeSWITCH! We're glad you're ditching a $120K system. We think you'll find FS is as powerful as any software out there right now. Here's a handy wiki page that will help you get the diagnosing skills you need: http://wiki.freeswitch.org/wiki/Reporting_Bugs I'd say first thing to do is capture the SIP traffic to see if there are any clues. A normal temporary failure doesn't give you a lot of detail. :) If you're new to SIP debugging then the best thing to do is to capture the SIP trace and put it in the pastebin. (http://pastebin.freeswitch.org) You can also join the IRC channel #freeswitch on irc.freenode.net http://irc.freenode.net and get some real-time help. There are some sharp folks in there, not the least of which are the three main FreeSWITCH developers. -MC Thank you. I think I did
Re: [Freeswitch-users] Newbie trying to setup Cisco 7940 phones
Peter J. Zandvoort pe...@cindyandpeter.com wrote: After looking at various asterisk distributions, SipX, 3CX and what-have-you, I've come to the conclusion that FreeSWITCH is by far the most advanced platform out there. Its architecture and performance is literally light years ahead of the rest and I have yet to come up with something that it can't do. But all that comes at a price: The learning curve is like scaling a brick wall. The most flexible and sophisticated tools tend to have this characteristic, the best solution to which is a supportive community and good documentation. FreeSWITCH has the community; the documentation is improving thanks to ongoing efforts to extend, clarify and enhance the wiki. ___ 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] Sipura Codec Problem
just an off-topic question but it concenns mass provissioning ... does anyone know if there is an open TR069 platform we can work on? T. On Wed, Nov 4, 2009 at 1:16 AM, Michael Collins m...@freeswitch.org wrote: On Tue, Nov 3, 2009 at 11:11 AM, Kristian Kielhofner kristian.kielhof...@gmail.com wrote: It appears that Tony has already added an option (amazing) BUT you should really be setup for central provisioning with an installed base that large... You'll eventually have issues that *NO* amount of Tony/FreeSWITCH magic can fix. Kristian is correct. Listen to him because he's familiar with having lots and lots of units out in the field. The bandage Tony applied will eventually wear off. The long-term solution is to treat the malady and not the symptom. I'm certain that members of the FS community could point you toward some resources to assist with central provisioning. -MC ___ 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 ___ 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
