Re: [Freeswitch-users] One more thing..
Are you doing this with the latest revision? You would have to supply more info like a console trace on debug level with siptrace enabled. On Feb 17, 2010 1:51 AM, "Doc" <ledoktre@meanie.us> wrote: Greetings one more time, I just remembered. I also ran into one other issue in my testing. When I route the incoming skype call to an extension (Sipura SPA 1001), it plays hold music on the callers side (good....), and when I pick up the extension, calling party drops, internal phone is left with a fast busy signal : 2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel [sofia/internal/sip:1001@10.24.72.12:5060] has been answered 2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use ptime 3 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121 sofia/internal/sip:1001@10.24.72.12:5060 has no read codec. First thing I notice is this unusual error about the ptime? and the one that hangs me up I think is the "sofia... has no read codec". I have been poking around my configuration - any suggestions? Thanks, Doc _______________________________________________ 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
Its the same bug the linksys SPA has... where the RTP time is set to 0.030 and an inbound call to the device doesn't 200ok with the right ptime. /b On Feb 17, 2010, at 7:36 AM, Anthony Minessale wrote:
Are you doing this with the latest revision? You would have to supply more info like a console trace on debug level with siptrace enabled.
Is there any configuration to change in the Sipura, any changes to FS configuration, or does it boil down to using different SIP hardware? -Thanks Brian West wrote:
Its the same bug the linksys SPA has... where the RTP time is set to 0.030 and an inbound call to the device doesn't 200ok with the right ptime.
/b
On Feb 17, 2010, at 7:36 AM, Anthony Minessale wrote:
Are you doing this with the latest revision? You would have to supply more info like a console trace on debug level with siptrace enabled.
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I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet Size) from 0.030 to 0.020, and the error went away. I was able to then test (and succeed) dialing from secondary skype user to skypiax, and have the call bridged automatically to one of my Sipura SPA-1001's. I could speak into the phone, and hear it come through skype (on my laptop) no problem. When I would talk on the laptop, I would get garble back on the phone, but that could be a sound issue on my laptop (my skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic). The interesting thing it did do, however, was eventually (within a minute?) It threw a couple of errors : 2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 And on a subsequent test, I received no audio (the above error rolling on the console), and within a matter of seconds, FS crashed with this : Segmentation fault (core dumped) I have another request open on the group actually for the above error, so I think it'd make sense to discuss further there (rather than duplicating material). Thanks! At any rate, the error for the missing codec, etc, seemed to be gone once I updated the RTP time in my ATA. Hope this helps someone!! Brian West wrote:
change the rtp time to 0.020 from 0.030
/b
keep updating, the maintainer of mod_skypeiax is adding new patches every few hours. You can also join irc on irc.freenode.net #freeswitch and #freeswitch-dev to interact with him live. One hint around here when we ask if you updated we work in 1 minute increments. On Wed, Feb 17, 2010 at 11:56 AM, Doc <ledoktre@meanie.us> wrote:
I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet Size) from 0.030 to 0.020, and the error went away. I was able to then test (and succeed) dialing from secondary skype user to skypiax, and have the call bridged automatically to one of my Sipura SPA-1001's. I could speak into the phone, and hear it come through skype (on my laptop) no problem. When I would talk on the laptop, I would get garble back on the phone, but that could be a sound issue on my laptop (my skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).
The interesting thing it did do, however, was eventually (within a minute?) It threw a couple of errors :
2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1
And on a subsequent test, I received no audio (the above error rolling on the console), and within a matter of seconds, FS crashed with this :
Segmentation fault (core dumped)
I have another request open on the group actually for the above error, so I think it'd make sense to discuss further there (rather than duplicating material).
Thanks! At any rate, the error for the missing codec, etc, seemed to be gone once I updated the RTP time in my ATA. Hope this helps someone!!
Brian West wrote:
change the rtp time to 0.020 from 0.030
/b
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900
I will update the software and report back. It it good to know that you think in "1 minute increments" - I figured mine was not too outdated ;-) -Thanks, Anthony Minessale wrote:
keep updating, the maintainer of mod_skypeiax is adding new patches every few hours. You can also join irc on irc.freenode.net <http://irc.freenode.net> #freeswitch and #freeswitch-dev to interact with him live.
One hint around here when we ask if you updated we work in 1 minute increments.
On Wed, Feb 17, 2010 at 11:56 AM, Doc <ledoktre@meanie.us <mailto:ledoktre@meanie.us>> wrote:
I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet Size) from 0.030 to 0.020, and the error went away. I was able to then test (and succeed) dialing from secondary skype user to skypiax, and have the call bridged automatically to one of my Sipura SPA-1001's. I could speak into the phone, and hear it come through skype (on my laptop) no problem. When I would talk on the laptop, I would get garble back on the phone, but that could be a sound issue on my laptop (my skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).
The interesting thing it did do, however, was eventually (within a minute?) It threw a couple of errors :
2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1
And on a subsequent test, I received no audio (the above error rolling on the console), and within a matter of seconds, FS crashed with this :
Segmentation fault (core dumped)
I have another request open on the group actually for the above error, so I think it'd make sense to discuss further there (rather than duplicating material).
Thanks! At any rate, the error for the missing codec, etc, seemed to be gone once I updated the RTP time in my ATA. Hope this helps someone!!
Brian West wrote: > change the rtp time to 0.020 from 0.030 > > /b
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-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 <http://iax:guest@conference.freeswitch.org/888> googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:+19193869900 ------------------------------------------------------------------------
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I have updated to "FreeSWITCH Version 1.0.trunk (16679)" which seems to have cleared up part # 2 of my question (issues with the errors being thrown on the screen during a call, and then subsequent calls failing). About the DTMF issues, for fun, I got onto Skype chat. I was told by the technician there that Skype does not support DTMF on SkypeIn -OR- Skype for SIP. He said sorry, but you won't get anywhere. I was also told they are not doing any blocking, that the reason its not working is probably due to all the transcoding that is taking place has degraded the signal too much. I guess I might try and see if there is a module available for a voice driven IVR? (any suggestions?) And on the rest of my first posting, any thoughts on why I must push a # sign when dialing an extension, etc, in FS? If I don't, I have to wait like 30 seconds. Also, the errors when I run startskype,sh (I am thinking must be no big deal) listed, are they any problem : expected keysym, got XF86KbdLightOnOff: line 70 of pc expected keysym, got XF86KbdBrightnessDown: line 71 of pc expected keysym, got XF86KbdBrightnessUp: line 72 of pc Could not init font path element /usr/share/fonts/X11/cyrillic, removing from list! Could not init font path element /var/lib/defoma/x-ttcidfont-conf.d/dirs/TrueType, removing from list! ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM null Thanks again you guys for all your help! Doc Anthony Minessale wrote:
keep updating, the maintainer of mod_skypeiax is adding new patches every few hours. You can also join irc on irc.freenode.net <http://irc.freenode.net> #freeswitch and #freeswitch-dev to interact with him live.
One hint around here when we ask if you updated we work in 1 minute increments.
On Wed, 17 Feb 2010 18:19 -0600, "Doc" <ledoktre@meanie.us> wrote:
About the DTMF issues, for fun, I got onto Skype chat. I was told by the technician there that Skype does not support DTMF on SkypeIn -OR- Skype for SIP. He said sorry, but you won't get anywhere.
Hi Doc, I can confirm that out-of-band DTMF signaling works for the Skype SIP product using a SkypeIn PSTN number.
From the Skype support site. https://support.skype.com/en/faq/FA10292/Do-you-support-DTMF-with-Skype-for-...
<quote> Do you support DTMF with Skype for SIP? Yes. Skype supports out-of-band DTMF signaling in accordance with the RFC 2833 standard. Details of RFC 2833 are located on the IETF website. We do *not* support in-band DTMF signaling. </quote> Regards, Scott Torr regards
Are you able to update your FS to the latest trunk? ("make current" in the FS src dir) Also, I notice that you are on 16619M - the "M" indicates a modification of some kind. Did you make any changes to the FS source? -MC On Wed, Feb 17, 2010 at 9:56 AM, Doc <ledoktre@meanie.us> wrote:
I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet Size) from 0.030 to 0.020, and the error went away. I was able to then test (and succeed) dialing from secondary skype user to skypiax, and have the call bridged automatically to one of my Sipura SPA-1001's. I could speak into the phone, and hear it come through skype (on my laptop) no problem. When I would talk on the laptop, I would get garble back on the phone, but that could be a sound issue on my laptop (my skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).
The interesting thing it did do, however, was eventually (within a minute?) It threw a couple of errors :
2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1
And on a subsequent test, I received no audio (the above error rolling on the console), and within a matter of seconds, FS crashed with this :
Segmentation fault (core dumped)
I have another request open on the group actually for the above error, so I think it'd make sense to discuss further there (rather than duplicating material).
Thanks! At any rate, the error for the missing codec, etc, seemed to be gone once I updated the RTP time in my ATA. Hope this helps someone!!
Brian West wrote:
change the rtp time to 0.020 from 0.030
/b
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I should be able to update to the latest trunk. Thanks for including the instructions. The only "modification" I did to the source was to edit the modules file and enable mod_skypiax. I will repost with results once I obtain them. -Thanks, Michael Collins wrote:
Are you able to update your FS to the latest trunk? ("make current" in the FS src dir) Also, I notice that you are on 16619M - the "M" indicates a modification of some kind. Did you make any changes to the FS source?
-MC
I have updated to "FreeSWITCH Version 1.0.trunk (16679)" per your instructions. My issues with this request all seem to be handled now. I had one or two things yet in my other posting, but I will respond there. THANK YOU !! Michael Collins wrote:
Are you able to update your FS to the latest trunk? ("make current" in the FS src dir) Also, I notice that you are on 16619M - the "M" indicates a modification of some kind. Did you make any changes to the FS source?
-MC
On Wed, Feb 17, 2010 at 9:56 AM, Doc <ledoktre@meanie.us <mailto:ledoktre@meanie.us>> wrote:
I did try changing the rtp time in my Sipura (advanced, sip, RTP Packet Size) from 0.030 to 0.020, and the error went away. I was able to then test (and succeed) dialing from secondary skype user to skypiax, and have the call bridged automatically to one of my Sipura SPA-1001's. I could speak into the phone, and hear it come through skype (on my laptop) no problem. When I would talk on the laptop, I would get garble back on the phone, but that could be a sound issue on my laptop (my skype sometimes does this with Pulse audio on my laptop, Ubuntu Karmic).
The interesting thing it did do, however, was eventually (within a minute?) It threw a couple of errors :
2010-02-17 11:47:34.007909 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 2010-02-17 11:47:34.027776 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1 2010-02-17 11:47:34.047643 [ERR] mod_skypiax.c:826 rev 16619M[(nil)|37 ][ERRORA 826 ][interface1][-1, 5,21] EXIT? sent=-1
And on a subsequent test, I received no audio (the above error rolling on the console), and within a matter of seconds, FS crashed with this :
Segmentation fault (core dumped)
I have another request open on the group actually for the above error, so I think it'd make sense to discuss further there (rather than duplicating material).
Thanks! At any rate, the error for the missing codec, etc, seemed to be gone once I updated the RTP time in my ATA. Hope this helps someone!!
Brian West wrote: > change the rtp time to 0.020 from 0.030 > > /b
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I need to be a little quicker on my replies :-) FS tells me "FreeSWITCH Version 1.0.trunk (16619M)". I installed it this past Saturday from SVN trunk as I recall. I can supply anything you need, but I am pretty new to FS and definately wet behind the ears. Thanks- Anthony Minessale wrote:
Are you doing this with the latest revision? You would have to supply more info like a console trace on debug level with siptrace enabled.
On Feb 17, 2010 1:51 AM, "Doc" <ledoktre@meanie.us <mailto:ledoktre@meanie.us>> wrote:
Greetings one more time,
I just remembered. I also ran into one other issue in my testing.
When I route the incoming skype call to an extension (Sipura SPA 1001), it plays hold music on the callers side (good....), and when I pick up the extension, calling party drops, internal phone is left with a fast busy signal :
2010-02-16 21:09:31.687204 [NOTICE] sofia.c:4690 Channel [sofia/internal/sip:1001@10.24.72.12:5060 <http://sip:1001@10.24.72.12:5060>] has been answered 2010-02-16 21:09:31.987714 [WARNING] mod_sofia.c:918 We were told to use ptime 3 but what they meant to say was 20 This issue has so far been identified to happen on the following broken platforms/devices: Linksys/Sipura aka Cisco ShoreTel Sonus/L3 We will try to fix it but some of the devices on this list are so broken who knows what will happen.. 2010-02-16 21:09:32.027243 [ERR] switch_core_io.c:121 sofia/internal/sip:1001@10.24.72.12:5060 <http://sip:1001@10.24.72.12:5060> has no read codec.
First thing I notice is this unusual error about the ptime? and the one that hangs me up I think is the "sofia... has no read codec". I have been poking around my configuration - any suggestions?
Thanks,
Doc
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participants (5)
-
Anthony Minessale
-
Brian West
-
Doc
-
Michael Collins
-
Scott Torr