Re: [asterisk-users] PRI got event HDLC Abort
is the card sharing irq? no. this the only card that uses IRQ 30 1b:00.0 Network controller: Digium, Inc. Device 1420 (rev 14) Subsystem: Device 0005: Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV+ VGASnoop- ParErr+ Stepping- SERR+ FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow TAbort- TAbort- MAbort- SERR- PERR- INTx- Latency: 64 (8000ns min, 32000ns max), Cache Line Size: 64 bytes Interrupt: pin A routed to IRQ 30 Region 0: Memory at 97a0 (32-bit, non-prefetchable) [size=32K] Kernel driver in use: wct4xxp is your system plugged directly into an outlet without ups? Please give us a complete lspci -vvv. Did you read this? http://alexrrr.blogspot.de/2007/10/solving-asterisks-hdlc-abort-issue.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] play wav file
I have an mp3 that is 128K, 44.1K stereo. I convert that to wave 16 bit, stereo, 44.1K The sound alike at this time. I want to play them (not just over my sound port) but through asterisk on select devices/machines that are also running asterisk over the Console/dsp. I converted the wave file to 8K, mono and it doesn't sound very good, I am also using 1.4.43 and ulaw,alaw,gsm allowed. What format will give me the best sounding output and how do I get that? Do I need somethink like g722? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play wav file
If you're going to stay with 1.4.X probably g722 would be best for you. If you work a while with SOX, you should end up with 8K files that sound almost as good as the 44K wav files. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, November 05, 2012 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] play wav file I have an mp3 that is 128K, 44.1K stereo. I convert that to wave 16 bit, stereo, 44.1K The sound alike at this time. I want to play them (not just over my sound port) but through asterisk on select devices/machines that are also running asterisk over the Console/dsp. I converted the wave file to 8K, mono and it doesn't sound very good, I am also using 1.4.43 and ulaw,alaw,gsm allowed. What format will give me the best sounding output and how do I get that? Do I need somethink like g722? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] play wav file
On Mon, Nov 5, 2012 at 10:52 AM, Jerry Geis ge...@pagestation.com wrote: I converted the wave file to 8K, mono and it doesn't sound very good, I am also using 1.4.43 and ulaw,alaw,gsm allowed. This has been covered just recently, try searching for mp3 on the mailing list. What format will give me the best sounding output and how do I get that? Do I need somethink like g722? Keep in mind that you are going to be using codecs and hardware that are optimized for speech, so anything that isn't speech is not going to sound good. In that case, best is really going to depend on what the content is and will probably require you to simply test all of the permutations and find the one that sounds the least bad. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USB FXS device
On 11/04/2012 04:17 AM, Andreas Sikkema wrote: Draytek Vigor2110Vn Sadly this doesn't seem to do OpenVPN, though it does several other flavors we might be able to support. Thanks for the tip! Will be looking into it. Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 11.0.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.0.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.0.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: * --- chan_sip: Fix a bug causing SIP reloads to remove all entries from the registry (Closes issue ASTERISK-20611. Reported by Alisher) * --- confbridge: Fix a bug which made conferences not record with AMI/CLI commands (Closes issue ASTERISK-20601. Reported by Vilius) * --- Fix an issue with res_http_websocket where the chan_sip WebSocket handler could not be registered. (Closes issue ASTERISK-20631. Reported by danjenkins) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.0.1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI got event HDLC Abort
-- Message: 6 Date: Mon, 5 Nov 2012 11:03:27 -0600 From: Danny Nicholas da...@debsinc.com Subject: Re: [asterisk-users] play wav file To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: 00e801cdbb77$79c701c0$6d550540$@debsinc.com Content-Type: text/plain; charset=us-ascii If you're going to stay with 1.4.X probably g722 would be best for you. If you work a while with SOX, you should end up with 8K files that sound almost as good as the 44K wav files. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis Sent: Monday, November 05, 2012 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] play wav file I have an mp3 that is 128K, 44.1K stereo. I convert that to wave 16 bit, stereo, 44.1K The sound alike at this time. I want to play them (not just over my sound port) but through asterisk on select devices/machines that are also running asterisk over the Console/dsp. I converted the wave file to 8K, mono and it doesn't sound very good, I am also using 1.4.43 and ulaw,alaw,gsm allowed. What format will give me the best sounding output and how do I get that? Do I need somethink like g722? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Message: 7 Date: Mon, 5 Nov 2012 11:04:36 -0600 From: Christopher Harrington ch...@acsdi.com Subject: Re: [asterisk-users] play wav file To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: cajlbxekhmmufgn9snuyctt8bxohwxcqqqocaswfcq7fqj1u...@mail.gmail.com Content-Type: text/plain; charset=utf-8 On Mon, Nov 5, 2012 at 10:52 AM, Jerry Geis ge...@pagestation.com wrote: I converted the wave file to 8K, mono and it doesn't sound very good, I am also using 1.4.43 and ulaw,alaw,gsm allowed. This has been covered just recently, try searching for mp3 on the mailing list. What format will give me the best sounding output and how do I get that? Do I need somethink like g722? Keep in mind that you are going to be using codecs and hardware that are optimized for speech, so anything that isn't speech is not going to sound good. In that case, best is really going to depend on what the content is and will probably require you to simply test all of the permutations and find the one that sounds the least bad. -- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20121105/a35675f5/attachment-0001.htm -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2010 - October 26-28 Washington, DC Register Now: http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 100, Issue 6 ** -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Support from Digium
On 11/4/2012 2:37 PM, Danny Dias wrote: Thanks Andrew, But i'm quite confuse with the following: *Q: Does Digium offer SLA guaranteed support for Asterisk?* *A:* Yes. Digium offers SLA guaranteed support, to SLA-entitled customers, for the Certified Asterisk branches. Digium does not offer SLA guaranteed support for other branches or releases. Just for Certify Versions of Asterisk? What does SLA means exactly? For example, if i install a FreePBX/Elastix (i'm not a good friend of these systems, but customers always ask for a web interface for management) to a customer, can i buy support from Digium for the Asterisk Release used? It would be nice to now the scope and limits of this support Thanks Digium offers a range of support options for Asterisk systems, regardless of whether you use a GUI (like FreePBX) or not. We do not provide support for the FreePBX software... but we can support Asterisk even with FreePBX in place. SLA stands for Service Level Agreement and is the highest tier of support. Since this is the asterisk-users list and not asterisk-biz I'll E-mail you directly for further discussion. asterisk-users is not the place for a discussion of commercial support options. Thanks, -- Rusty Newton Digium, Inc | Open Source Community Support Manager Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to create channel of type 'DAHDI' (cause 17 - User busy)
On 11/2/2012 4:58 AM, Harish Mandowara wrote: Hi, I have 6 Red FXO with TDM2400p in my PC. I have install asterisk and dahdi driver. Scenario is The quickest way to get pointed in the right direction would be to contact Digium support since the issue involves your TDM2400P. Be sure you have the serial number of your card when you call (printed on the card itself, and included with the documentation you received upon purchase). http://www1.digium.com/en/support/contact Please don't cross-post across the lists. Thanks, -- Rusty Newton Digium, Inc | Open Source Community Support Manager Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax Configuration
What is the best way for me to setup Fax Capability with VOIP only. I have a Asterisk Server hosted on the internet without a modem. I'm using Flowroute, which is working awesome, for VOIP calls. I only have a SIP Phone at home and two Printer/Scanner/Fax Printers. I'm not sure which Fax Addons or Extensions I should use for Asterisk. I'd like it to Self Detect on any line. I also am not sure what or how I can connect a Network Only Printer/Scanner/Fax Machine at home to it. It has a Telephone Jack but I'm only using VOIP. I'm pretty advanced with Asterisk now and can figure things out..I would just like some advice and direction before I get started. Oh, one more thing. Is there any way to Route the Faxes to different folders (extensions) because I have End Users with Phone + Extensions when you call in. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336 Phone/Fax (855) 760-COOP (2667) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI got event HDLC Abort
On 11/5/12 11:59 AM, Vincent Swart wrote: You're HDLC error is evident of timing slips. Use cat /proc/dahdi/1 or 2 or 3 aha.. it does have timing slips... Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) B8ZS/ESF ClockSource CRC4 error count: 6864 E-bit error count: 27603 IRQ misses: 1 Timing slips: 1459 1 TE4/0/1/1 Clear (In use) (EC: VPMOCT128 - INACTIVE) 2 TE4/0/1/2 Clear (In use) (EC: VPMOCT128 - INACTIVE) 3 TE4/0/1/3 Clear (In use) (EC: VPMOCT128 - INACTIVE) 4 TE4/0/1/4 Clear (In use) (EC: VPMOCT128 - INACTIVE) 5 TE4/0/1/5 Clear (In use) (EC: VPMOCT128 - INACTIVE) 6 TE4/0/1/6 Clear (In use) (EC: VPMOCT128 - INACTIVE) 7 TE4/0/1/7 Clear (In use) (EC: VPMOCT128 - INACTIVE) 8 TE4/0/1/8 Clear (In use) (EC: VPMOCT128 - INACTIVE) 9 TE4/0/1/9 Clear (In use) (EC: VPMOCT128 - INACTIVE) 10 TE4/0/1/10 Clear (In use) (EC: VPMOCT128 - INACTIVE) 11 TE4/0/1/11 Clear (In use) (EC: VPMOCT128 - INACTIVE) 12 TE4/0/1/12 Clear (In use) (EC: VPMOCT128 - INACTIVE) 13 TE4/0/1/13 Clear (In use) (EC: VPMOCT128 - INACTIVE) 14 TE4/0/1/14 Clear (In use) (EC: VPMOCT128 - INACTIVE) 15 TE4/0/1/15 Clear (In use) (EC: VPMOCT128 - INACTIVE) 16 TE4/0/1/16 Clear (In use) (EC: VPMOCT128 - INACTIVE) 17 TE4/0/1/17 Clear (In use) (EC: VPMOCT128 - INACTIVE) 18 TE4/0/1/18 Clear (In use) (EC: VPMOCT128 - INACTIVE) 19 TE4/0/1/19 Clear (In use) (EC: VPMOCT128 - INACTIVE) 20 TE4/0/1/20 Clear (In use) (EC: VPMOCT128 - INACTIVE) 21 TE4/0/1/21 Clear (In use) (EC: VPMOCT128 - INACTIVE) 22 TE4/0/1/22 Clear (In use) (EC: VPMOCT128 - INACTIVE) 23 TE4/0/1/23 Clear (In use) (EC: VPMOCT128 - INACTIVE) 24 TE4/0/1/24 HDLCFCS (In use) (EC: VPMOCT128 - INACTIVE) Also cat /proc /interrupts however i don't see any interrupt conflicts.. maybe i should try manually assign CPU affinity on that IRQ? CPU0 CPU1 CPU2 CPU3 0: 2108 0 0 0 IO-APIC-edge timer 1: 0 0 0 0 IO-APIC-edge i8042 8: 1 0 0 0 IO-APIC-edge rtc0 9: 0 0 0 0 IO-APIC-fasteoi acpi 14: 89 0 0 0 IO-APIC-edge ata_piix 15: 0 0 0 0 IO-APIC-edge ata_piix 16: 608555 0 0 0 IO-APIC-fasteoi megasas 17: 51 0 0 0 IO-APIC-fasteoi ehci_hcd:usb2, uhci_hcd:usb3, uhci_hcd:usb5 18: 0 0 0 0 IO-APIC-fasteoi uhci_hcd:usb4, uhci_hcd:usb6 19: 0 0 0 0 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb7 21: 0 0 0 0 IO-APIC-fasteoi ata_piix 30: 604673256 0 0 0 IO-APIC-fasteoi wct4xxp 54: 3 0 0 0 PCI-MSI-edge ioat-msix 55: 3 0 0 0 PCI-MSI-edge ioat-msix 56: 3 0 0 0 PCI-MSI-edge ioat-msix 57: 3 0 0 0 PCI-MSI-edge ioat-msix 58: 3 0 0 0 PCI-MSI-edge ioat-msix 59: 3 0 0 0 PCI-MSI-edge ioat-msix 60: 3 0 0 0 PCI-MSI-edge ioat-msix 61: 3 0 0 0 PCI-MSI-edge ioat-msix 62: 772684 0 0 0 PCI-MSI-edge eth0-0 63: 368866 0 0 0 PCI-MSI-edge eth0-1 64: 105367 0 0 0 PCI-MSI-edge eth0-2 65: 0 0 0 0 PCI-MSI-edge eth0-3 66: 0 0 0 0 PCI-MSI-edge eth0-4 71: 22558707 0 0 0 PCI-MSI-edge eth1-0 72: 15994275 0 0 0 PCI-MSI-edge eth1-1 73: 24318397 0 0 0 PCI-MSI-edge eth1-2 74: 12812423 0 0 0 PCI-MSI-edge eth1-3 75: 11109627 0 0 0 PCI-MSI-edge eth1-4 NMI: 0 0 0 0 Non-maskable interrupts LOC: 50455701 61286848 31629357 13702410 Local timer interrupts SPU: 0 0 0 0 Spurious interrupts PMI: 0 0 0 0 Performance monitoring interrupts PND: 0 0 0 0 Performance pending work RES: 7017 18000 5306 1944 Rescheduling interrupts CAL:
Re: [asterisk-users] Fax Configuration
On 11/05/2012 04:18 PM, Roy Abshire wrote: What is the best way for me to setup Fax Capability with VOIP only. You can use T.38, perhaps, if your VoIP provider supports it and you can get it working. But unless you need faxes to go through the telephony system (i.e. you have fax machines hooked up to FXS ports) I'd recommend using an on-line fax service such as Mainpine's instead of trying to do fax over VoIP. Fax over internet-strewn SIP generally isn't going to work very well. I have a Asterisk Server hosted on the internet without a modem. I'm using Flowroute, which is working awesome, for VOIP calls. I only have a SIP Phone at home and two Printer/Scanner/Fax Printers. So on your MFP you'll scan it instead of using the system's fax capability, and then fax it through the online service. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Configuration
Roy, Many will say that it all depends on your provider supporting T.38, and that you should forget it otherwise. My practical experience shows otherwise. I am able to receive faxes on SIP lines pretty reliably with no T.38 support. The biggest issue for me is CED tones detection. If CED is detected then fax reception goes on with no problems. I use this setup with both Asterisk receiving a fax and then e-mailing it to me as a PDF attachment and with a call being forwarded to a fax machine extension which in turn is connected to a Motorola ATA over another SIP connection. I use FreePBX to set all that up as they provide pretty easy fax setup. -Vladimir On 11/5/2012 6:18 PM, Roy Abshire wrote: What is the best way for me to setup Fax Capability with VOIP only. I have a Asterisk Server hosted on the internet without a modem. I'm using Flowroute, which is working awesome, for VOIP calls. I only have a SIP Phone at home and two Printer/Scanner/Fax Printers. I'm not sure which Fax Addons or Extensions I should use for Asterisk. I'd like it to Self Detect on any line. I also am not sure what or how I can connect a Network Only Printer/Scanner/Fax Machine at home to it. It has a Telephone Jack but I'm only using VOIP. I'm pretty advanced with Asterisk now and can figure things out..I would just like some advice and direction before I get started. Oh, one more thing. Is there any way to Route the Faxes to different folders (extensions) because I have End Users with Phone + Extensions when you call in. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Google Voice and back (chan_motif)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Today I started to experiment with Google Voice and Asterisk-11.0.1. Following the instructions on the wiki (https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google), I was able to make / receive calls quite easily with a single account on asterisk. Then I tried to add a second Google Voice account to Asterisk, and make calls between accounts. I defined a second connection in xmpp.conf, a second account in chan_motif (see relevant configuration below). I'm getting the following error: ERROR[28651][C-0002]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session (see full log below) Should I open a bug report or did I make an mistake in configuration? motif.conf: - --- [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw connection=google-cathy ; - xmpp.conf xmpp.conf: - -- [google-jd] type=client serverhost=talk.google.com username=jeandenis.gir...@gmail.com secret=xx priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Disponible - GMT-10 ! timeout=5 [google-cathy] type=client serverhost=talk.google.com username=cathy.fou...@gmail.com secret= priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Disponible - GMT-10 ! timeout=5 extensions.conf: - [incoming-motif] exten = s,1,NoOp() same = n,Wait(1) same = n,Answer() same = n,SendDTMF(1) same = n,Dial(SIP/FYJmmzJ3,20) call log: - - == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [72@i9PuqEcv:1] Dial(SIP/i9PuqEcv-0002, Motif/google-jd/cathy.fou...@gmail.com,,r) in new stack --- XMPP sent to 'google-jd' --- iq from='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' to='cathy.fou...@gmail.com/asterisk-xD2C13566' type='set' id='o'jingle action='session-initiate' sid='7e44df781ce623b6' xmlns='urn:xmpp:jingle:1' initiator='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06'content creator='initiator' name='audio'description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'payload-type id='110' name='speex' channels='1' clockrate='8000'/payload-type id='0' name='PCMU' channels='1' clockrate='8000'/payload-type id='9' name='G722' channels='1' clockrate='8000'/payload-type id='8' name='PCMA' channels='1' clockrate='8000'/payload-type id='101' name='telephone-event' channels='1' clockrate='8000'//descriptiontransport xmlns='urn:xmpp:jingle:transports:ice-udp:1'//content/jingle/iq - -- Called Motif/google-jd/cathy.fou...@gmail.com --- XMPP received from 'google-jd' --- - --- XMPP received from 'google-cathy' --- iq from=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 to=cathy.fou...@gmail.com/asterisk-xD2C13566 type=set id=ojingle action=session-initiate sid=7e44df781ce623b6 initiator=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 xmlns=urn:xmpp:jingle:1content creator=initiator name=audiodescription media=audio xmlns=urn:xmpp:jingle:apps:rtp:1payload-type id=110 name=speex channels=1 clockrate=8000/payload-type id=0 name=PCMU channels=1 clockrate=8000/payload-type id=9 name=G722 channels=1 clockrate=8000/payload-type id=8 name=PCMA channels=1 clockrate=8000/payload-type id=101 name=telephone-event channels=1 clockrate=8000//descriptiontransport xmlns=urn:xmpp:jingle:transports:ice-udp:1//content/jingle/iq - --- XMPP sent to 'google-cathy' --- iq type='result' from='cathy.fou...@gmail.com/asterisk-xD2C13566' to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' id='o'/ - --- XMPP sent to 'google-cathy' --- iq from='cathy.fou...@gmail.com/asterisk-xD2C13566' to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' type='set' id='j'jingle action='transport-info' sid='7e44df781ce623b6' xmlns='urn:xmpp:jingle:1'content creator='responder' name='audio'transport xmlns='urn:xmpp:jingle:transports:ice-udp:1' pwd='4b4001b575f3c7b824e14d9436d5f466' ufrag='6c28e0a07a5269e82ee313d916a046f7'candidate component='1' foundation='583375015' generation='0' id='0a86' ip='192.168.1.1' port='16384' priority='2130706431' protocol='udp' type='host'/candidate component='1' foundation='583378294' generation='0' id='3c7f' ip='192.168.0.10' port='16384' priority='2130706431' protocol='udp' type='host'/candidate component='1' foundation='192809686' generation='0' id='85cc' ip='123.50.122.114' port='16384' priority='2130706431' protocol='udp' type='host'/candidate component='2' foundation='583375015' generation='0' id='cc6e' ip='192.168.1.1' port='16385' priority='2130706430' protocol='udp' type='host'/candidate component='2' foundation='583378294' generation='0' id='5cb8' ip='192.168.0.10' port='16385'
Re: [asterisk-users] Google Voice and back (chan_motif)
Try adding transport=google-v1 to motif.conf [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-cathy ; - xmpp.conf On 11/05/2012 08:35 PM, Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Today I started to experiment with Google Voice and Asterisk-11.0.1. Following the instructions on the wiki (https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google), I was able to make / receive calls quite easily with a single account on asterisk. Then I tried to add a second Google Voice account to Asterisk, and make calls between accounts. I defined a second connection in xmpp.conf, a second account in chan_motif (see relevant configuration below). I'm getting the following error: ERROR[28651][C-0002]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session (see full log below) Should I open a bug report or did I make an mistake in configuration? motif.conf: - --- [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw connection=google-cathy ; - xmpp.conf xmpp.conf: - -- [google-jd] type=client serverhost=talk.google.com username=jeandenis.gir...@gmail.com secret=xx priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Disponible - GMT-10 ! timeout=5 [google-cathy] type=client serverhost=talk.google.com username=cathy.fou...@gmail.com secret= priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Disponible - GMT-10 ! timeout=5 extensions.conf: - [incoming-motif] exten = s,1,NoOp() same = n,Wait(1) same = n,Answer() same = n,SendDTMF(1) same = n,Dial(SIP/FYJmmzJ3,20) call log: - - == Using SIP VIDEO TOS bits 136 == Using SIP VIDEO CoS mark 6 == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Executing [72@i9PuqEcv:1] Dial(SIP/i9PuqEcv-0002, Motif/google-jd/cathy.fou...@gmail.com,,r) in new stack --- XMPP sent to 'google-jd' --- iq from='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' to='cathy.fou...@gmail.com/asterisk-xD2C13566' type='set' id='o'jingle action='session-initiate' sid='7e44df781ce623b6' xmlns='urn:xmpp:jingle:1' initiator='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06'content creator='initiator' name='audio'description xmlns='urn:xmpp:jingle:apps:rtp:1' media='audio'payload-type id='110' name='speex' channels='1' clockrate='8000'/payload-type id='0' name='PCMU' channels='1' clockrate='8000'/payload-type id='9' name='G722' channels='1' clockrate='8000'/payload-type id='8' name='PCMA' channels='1' clockrate='8000'/payload-type id='101' name='telephone-event' channels='1' clockrate='8000'//descriptiontransport xmlns='urn:xmpp:jingle:transports:ice-udp:1'//content/jingle/iq - -- Called Motif/google-jd/cathy.fou...@gmail.com --- XMPP received from 'google-jd' --- - --- XMPP received from 'google-cathy' --- iq from=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 to=cathy.fou...@gmail.com/asterisk-xD2C13566 type=set id=ojingle action=session-initiate sid=7e44df781ce623b6 initiator=jeandenis.gir...@gmail.com/asterisk-xBEE0DF06 xmlns=urn:xmpp:jingle:1content creator=initiator name=audiodescription media=audio xmlns=urn:xmpp:jingle:apps:rtp:1payload-type id=110 name=speex channels=1 clockrate=8000/payload-type id=0 name=PCMU channels=1 clockrate=8000/payload-type id=9 name=G722 channels=1 clockrate=8000/payload-type id=8 name=PCMA channels=1 clockrate=8000/payload-type id=101 name=telephone-event channels=1 clockrate=8000//descriptiontransport xmlns=urn:xmpp:jingle:transports:ice-udp:1//content/jingle/iq - --- XMPP sent to 'google-cathy' --- iq type='result' from='cathy.fou...@gmail.com/asterisk-xD2C13566' to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' id='o'/ - --- XMPP sent to 'google-cathy' --- iq from='cathy.fou...@gmail.com/asterisk-xD2C13566' to='jeandenis.gir...@gmail.com/asterisk-xBEE0DF06' type='set' id='j'jingle action='transport-info' sid='7e44df781ce623b6' xmlns='urn:xmpp:jingle:1'content creator='responder' name='audio'transport xmlns='urn:xmpp:jingle:transports:ice-udp:1' pwd='4b4001b575f3c7b824e14d9436d5f466' ufrag='6c28e0a07a5269e82ee313d916a046f7'candidate component='1' foundation='583375015' generation='0' id='0a86' ip='192.168.1.1' port='16384' priority='2130706431' protocol='udp' type='host'/candidate component='1' foundation='583378294' generation='0' id='3c7f' ip='192.168.0.10' port='16384' priority='2130706431'
Re: [asterisk-users] Google Voice and back (chan_motif)
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 05/11/2012 18:55, Co-op Vacation Rentals a écrit : Try adding transport=google-v1 to motif.conf [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-cathy ; - xmpp.conf Thanks for your reply, unfortunately that makes no difference, I still get: [Nov 5 19:45:16] ERROR[30664][C-0005]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session '14ec70fb484b5700' Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlCYpNoACgkQuu7Rv+oOo/imrgCgrDUi0VdhCbspzA7SUtFQWpDK iEAAn3X5x/eX96eSRj8PsXqpk4SYFpA5 =98GL -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice and back (chan_motif)
Here are my settings that work. I can make incoming and outgoing calls. Compare my settings with yours. Also make sure your firewall is open for port 5222 and 5060 and your RTP port range. #rtp.conf [general] icesupport=yes rtpstart=15000 rtpend=2 #motif.conf [default](!) disallow=all allow=alaw allow=ulaw allow=h264 transport=google-v1 context=incoming [asterisk](default) connection=asterisk [coopvr](default) connection=coopvr #xmpp.con [asterisk] type=client serverhost=talk.google.com username=coopaster...@gmail.com secret=xx priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Asterisk Server timeout=5 [coopvr] type=client serverhost=talk.google.com username=coo...@gmail.com secret=xx priority=1 port=5222 usetls=yes usesasl=yes status=available statusmessage=Asterisk Server timeout=5 On 11/05/2012 09:49 PM, Jean-Denis Girard wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 05/11/2012 18:55, Co-op Vacation Rentals a écrit : Try adding transport=google-v1 to motif.conf [google-jd] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-jd ; - xmpp.conf [google-cathy] context=incoming-motif disallow=all allow=speex allow=ulaw allow=g722 allow=h264 allow=alaw *transport=google-v1* connection=google-cathy ; - xmpp.conf Thanks for your reply, unfortunately that makes no difference, I still get: [Nov 5 19:45:16] ERROR[30664][C-0005]: chan_motif.c:1971 jingle_interpret_ice_udp_transport: Incomplete ICE-UDP candidate received on session '14ec70fb484b5700' Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAlCYpNoACgkQuu7Rv+oOo/imrgCgrDUi0VdhCbspzA7SUtFQWpDK iEAAn3X5x/eX96eSRj8PsXqpk4SYFpA5 =98GL -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roy Abshire Co-op Vacation Rentals 15218 Summit Ave Suite 300-354 Fontana, CA 92336 (855) 760-COOP (4667) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
