Re: [asterisk-users] asterisk-users Digest, Vol 75, Issue 7
Hi, I cannot get asterisk to start again after the g729 install failed. kindly advise what's the problem. Thank's -- _ -- 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] Polycom getting DCHP address from wrong VLAN
Hello, I have been tearing my hair out on this issue for 2 days, any help would be appreciated. We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch There are two VLANs, 1(data) 50(VoIP). When Polycoms are connected to the switch with VLAN 50 hard coded in the config they grab a DHCP address from VLAN 1, the PVID for the switch port. The ports have membership in VLAN 1 as the PVID and VLAN 50 as tagged traffic. I know the VoIP DHCP server is working because if I change a port to have a PVID of 50 any device gets the address from the VoIP DHCP server. I have tried the ports as 'general' and 'trunk' with no success. Any help would be greatly appreciated, I don't have much hair left! -- -Thermal -- _ -- 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] 401 Unauthorized with Snom but not with Zoiper softphone
On 10/07/2010 06:50 PM, Daniel Tryba wrote: On Thu, Oct 07, 2010 at 02:57:27PM +0200, Jonas Kellens wrote: nat=yes is set as a global parameter and also in the realtime MySQL sip_buddies database I have for every peer nat=yes. I then find it very strange that when placing these Snom phones in my environment (for configuration) work very well, and then when I hook them up at the site there is trouble with nat. I'm also behind nat here... I have never seen this problem before with Snom M3 and different routers (Linux/Cisco or stupid SpeedTouches) without any connection NAT helpers for SIP enabled. I'd say you should try the difference values for nat to see if one works with the NAT gateway or use STUN like suggested elsewhere. What STUN-server can I use then ?! Asterisk is no STUN-server I guess and is there then something like a public STUN-server ?! I have never experienced problems with NAT... The different options for nat are : nat=yes|no|never|route Correct ?! Some extra information : when using TELNET to my public Asterisk server on port 5060, then there is no response. This means that the answer is blocked somewhere, right ?! Maybe on ISP-level ?! Is that plausible ?! Jonas. -- _ -- 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] 401 Unauthorized with Snom but not with Zoiper softphone
Hello, there is a really great difference in the Via-header of the REGISTER-message between the Zoiper and the Snom. Also the Zoiper has a Contact-header, and the Snom REGISTER has not... Snom : REGISTER sip:sip.domain.tld SIP/2.0 _*Via: SIP/2.0/UDP 192.168.114.200:2049;branch=z9hG4bK-p4ayhthezdr6;rport*_ From: sip:te...@sip.domain.tld;tag=d85nzjmkpk To: sip:te...@sip.domain.tld Call-ID: 3c26701f88d8-6i37fwkca22u CSeq: 9 REGISTER Max-Forwards: 70 Contact: sip:te...@192.168.114.200:2049;line=c38yvjnm;reg-id=1;q=1.0;+sip.instance=urn:uuid:0c5bc641-5821-4562-a514-f9e81d97c118;audio;mobility=fixed;duplex=full;description=snom320;actor=principal;events=dialog;methods=INVITE,ACK,CANCEL,BYE,REFER,OPTIONS,NOTIFY,SUBSCRIBE,PRACK,MESSAGE,INFO User-Agent: snom320/8.4.18 Allow-Events: dialog X-Real-IP: 192.168.114.200 Supported: path, gruu Expires: 3600 Content-Length: 0 SIP/2.0 401 Unauthorized _*Via: SIP/2.0/UDP 192.168.114.200:2049;branch=z9hG4bK-p4ayhthezdr6;received=public_ip;rport=64646*_ From: sip:te...@sip.domain.tld;tag=d85nzjmkpk To: sip:te...@sip.domain.tld;tag=as21294ace Call-ID: 3c26701f88d8-6i37fwkca22u CSeq: 9 REGISTER Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=domain.tld, nonce=120e4f10 Content-Length: 0 Zoiper : REGISTER sip:sip.domain.tld;transport=UDP SIP/2.0 _*Via: SIP/2.0/UDP public_ip:59096;branch=z9hG4bK-d8754z-83ebdbc32d0fb2e0-1---d8754z-;rport*_ Max-Forwards: 70 _*Contact: sip:te...@public_ip:59096;rinstance=d6dc257201b6ffa2;transport=UDP*_ To: sip:te...@sip.domain.tld;transport=UDP From: sip:te...@sip.domain.tld;transport=UDP;tag=5a50ca11 Call-ID: NjY2MmY3NjRkNDRlOWJhOGQ2NzgyNzg4Y2M5ZGFlZjE. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper rev.7797 Allow-Events: presence, kpml Content-Length: 0 SIP/2.0 401 Unauthorized _*Via: SIP/2.0/UDP public_ip:59096;branch=z9hG4bK-d8754z-83ebdbc32d0fb2e0-1---d8754z-;received=public_ip;rport=59095*_ From: sip:te...@sip.domain.tld;transport=UDP;tag=5a50ca11 To: sip:te...@sip.domain.tld;transport=UDP;tag=as1640cd46 Call-ID: NjY2MmY3NjRkNDRlOWJhOGQ2NzgyNzg4Y2M5ZGFlZjE. CSeq: 1 REGISTER Server: Asterisk PBX 1.6.2.10 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=domain.tld, nonce=08277195 Content-Length: 0 Is that why Asterisk is not able to send the 401 or the 200 ?? Anyone knows if there is a special setting in the Snom 320 or Asterisk itself to overcome this issue ?! Jonas. -- _ -- 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 router
On 7 Oct 2010, at 23:57, steve casto wrote: A Crisco RVS4000 installed now has real problems with Sip, one-way audio and throughput not up to the WAN speed. ALG? (Assuming you mean Cisco..) -- _ -- 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] Voice quality assessment in Asterisk
Hi, How do you typically test voice quality in Asterisk? For example if you like to do load testing, or monitor voice quality and get notified if certain calls had bad quality for proactive maintenance? Thank you! Best Regards, Sevana Oy http://www.sevana.fi-- _ -- 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] Voice quality assessment in Asterisk
The professional way is to do a series of test calls, play a reference file and record the audio at the incoming side. You then use both files to calculate a MOS score. This method is used by telco's to do quality checks. https://secure.wikimedia.org/wikipedia/en/wiki/Mean_opinion_score http://voip.about.com/od/voipbasics/a/MOS.htm Bert On 08/10/2010 11:12, Sevana Oy wrote: Hi, How do you typically test voice quality in Asterisk? For example if you like to do load testing, or monitor voice quality and get notified if certain calls had bad quality for proactive maintenance? Thank you! Best Regards, Sevana Oy http://www.sevana.fi -- _ -- 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] Voice quality assessment in Asterisk
On Fri, Oct 08, 2010 at 02:24:11PM +0200, Bert Van Kets wrote: The professional way is to do a series of test calls, play a reference file and record the audio at the incoming side. You then use both files to calculate a MOS score. This method is used by telco's to do quality checks. Take a look at the website mentioned in GPs post. He/they already know this, I guess it is a fishing expedition for competitors :) We don't do the test calls method, but use inline probes (Fluke ACEs) that analyze all traffic and give a MOS score to SIP calls and save network statistics per call (can be retrieved from the RTCP reports in asterisk). These probes and the analyzer software aren't bug free and perfect but give a good indication of all historic calls. Once a problem is spotted we move to test calls to trace the problem. -- Daniel Tryba -- _ -- 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] Voice quality assessment in Asterisk
Those boxes run around $50k USD, I've only seen them once back in the late 1990s. At work for customer consulting we have very expensive site licenses for Prognosis IPT Assessor which generate great looking pdf reports. We also use Cisco IOS IP SLA however it doesn't have a reporting mechanism. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bert Van Kets Sent: Friday, October 08, 2010 8:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Voice quality assessment in Asterisk The professional way is to do a series of test calls, play a reference file and record the audio at the incoming side. You then use both files to calculate a MOS score. This method is used by telco's to do quality checks. https://secure.wikimedia.org/wikipedia/en/wiki/Mean_opinion_score http://voip.about.com/od/voipbasics/a/MOS.htm Bert On 08/10/2010 11:12, Sevana Oy wrote: Hi, How do you typically test voice quality in Asterisk? For example if you like to do load testing, or monitor voice quality and get notified if certain calls had bad quality for proactive maintenance? Thank you! Best Regards, Sevana Oy http://www.sevana.fi - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is for use by the designated addressee(s) named above only. If you are not the intended addressee, you are hereby notified that you have received this communication in error and that any use or reproduction of this email or its contents is strictly prohibited and may be unlawful. If you have received this communication in error, please notify us immediately by replying to this message and deleting it from your computer. Thank you.-- _ -- 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] Dahdi error
You´re right!! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Fri, 8 Oct 2010 00:16:58 -0500 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dahdi error On 10/7/10 2:07 PM, Flavio Miranda wrote: asterisk:/etc/asterisk# /etc/init.d/dahdi start Loading DAHDI hardware modules: FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Device or resource busy wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done wcfxo: done wctdm: done wcb4xxp: error wctc4xxp: done xpp_usb: done Error: missing /dev/dahdi! Best guess based on the information you provided: zaptel was installed on this machine and is already loaded and registered major number 196. That would explain both the Device or resource busy error, and the fact that dahdi failed to load, yet most of the board drivers appear to have loaded (since the zaptel ones probably loaded up and the wcb4xxp driver did not load). -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA 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 -- _ -- 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] Voice quality assessment in Asterisk
One quick clarification please... With Fluke ACEs you measure MOS according G.107, E-model, right? Thanks a lot to all who replied and will reply! - Original Message - From: Daniel Tryba dan...@tryba.nl To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, October 08, 2010 4:41 PM Subject: Re: [asterisk-users] Voice quality assessment in Asterisk On Fri, Oct 08, 2010 at 02:24:11PM +0200, Bert Van Kets wrote: The professional way is to do a series of test calls, play a reference file and record the audio at the incoming side. You then use both files to calculate a MOS score. This method is used by telco's to do quality checks. Take a look at the website mentioned in GPs post. He/they already know this, I guess it is a fishing expedition for competitors :) We don't do the test calls method, but use inline probes (Fluke ACEs) that analyze all traffic and give a MOS score to SIP calls and save network statistics per call (can be retrieved from the RTCP reports in asterisk). These probes and the analyzer software aren't bug free and perfect but give a good indication of all historic calls. Once a problem is spotted we move to test calls to trace the problem. -- Daniel Tryba -- _ -- 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
[asterisk-users] How to use Atxfer in AMI
Hi, I'm trying to make a attended transfer through AMI. I though i could use Atxfer, and it seems ok, but nothing happens. And I can't find any how-to or description on how to do this. What more do I have to do to make this work? In Asterisk Call Manager: Action: Atxfer Channel: SIP/36-xx Exten: 33 Priority: 1 Context: Phone Response: Success Message: Atxfer successfully queued Best regards, Kent Varmedal -- _ -- 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] Weird stalling of playback on IAX2 channels on 1.8 svn
I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- 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] Weird stalling of playback on IAX2 channels on 1.8svn
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Friday, October 08, 2010 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8svn I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk Are both Asterisk's 1.8? I had unhappy results doing IAX between 1.4 and 1.6 (1.8 is built on 1.6???) -- _ -- 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] Weird stalling of playback on IAX2 channels on 1.8svn
On 8 Oct 2010, at 15:37, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Friday, October 08, 2010 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8svn I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk Are both Asterisk's 1.8? I had unhappy results doing IAX between 1.4 and 1.6 (1.8 is built on 1.6???) The far end is our voip supplier's asterisk - no Idea what version. But it also happens when talking to our Java IAX stack which isn't asterisk at all, so it isn't specific to a particular asterisk :-) What's more, if a call makes it past the announcement and gets bridged, it works fine. I've had several half hour calls through it. So it seems to me that it is an interaction between playback and iax2. T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- 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] Polycom getting DCHP address from wrong VLAN
That switch doesn't seem to support CDP, so the Polycom phone has no way of figuring out which VLAN to tag itself to automagically. It will grab the primary VLAN unless you specify otherwise in the phone's setup. On boot of the phone, go into setup, default password 456, there's an option in there to specify the VLAN ID. You can force your VLAN 50. Should bring you up. If your switch had done CDP (such as some higher end switches), the phone tells the switch Im a phone and the switch has a voice-vlan configuration which would then push that VLAN to the Polycom. We did this kind of setup in a large scale (100+) installation. Bests, Seb --- Sebastien Thomas Amplisys Inc. - Digital Telephony Integration Specialists T: 514.225.4141 x222 F: 514.225.4162 TF: 1-877-AMPLISYS On 2010-10-08, at 2:37 AM, Thermal Wetland wrote: Hello, I have been tearing my hair out on this issue for 2 days, any help would be appreciated. We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch There are two VLANs, 1(data) 50(VoIP). When Polycoms are connected to the switch with VLAN 50 hard coded in the config they grab a DHCP address from VLAN 1, the PVID for the switch port. The ports have membership in VLAN 1 as the PVID and VLAN 50 as tagged traffic. I know the VoIP DHCP server is working because if I change a port to have a PVID of 50 any device gets the address from the VoIP DHCP server. I have tried the ports as 'general' and 'trunk' with no success. Any help would be greatly appreciated, I don't have much hair left! -- -Thermal -- _ -- 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] Weird stalling of playback on IAX2 channels on 1.8 svn
Tim I am actually seeing this on a 1.6.2.13 box as well. For some reason durring prompt playbacks they some times stall mid file. The call stays up but no audio comes in. Bryant From: Tim Panton t...@westhawk.co.uk Sent: Friday, October 08, 2010 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8 svn I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- 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] Weird stalling of playback on IAX2 channels on 1.8svn
On 8 Oct 2010, at 15:37, Danny Nicholas wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Panton Sent: Friday, October 08, 2010 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8svn I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk Are both Asterisk's 1.8? I had unhappy results doing IAX between 1.4 and 1.6 (1.8 is built on 1.6???) The far end is our voip supplier's asterisk - no Idea what version. But it also happens when talking to our Java IAX stack which isn't asterisk at all, so it isn't specific to a particular asterisk :-) What's more, if a call makes it past the announcement and gets bridged, it works fine. I've had several half hour calls through it. So it seems to me that it is an interaction between playback and iax2. T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk Tim my issues with 1.6.13 have been on sip Bryant -- _ -- 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] Weird stalling of playback on IAX2 channels on 1.8 svn
On 8 Oct 2010, at 16:03, Bryant Zimmerman wrote: Tim I am actually seeing this on a 1.6.2.13 box as well. For some reason durring prompt playbacks they some times stall mid file. The call stays up but no audio comes in. Bryant From: Tim Panton t...@westhawk.co.uk Sent: Friday, October 08, 2010 10:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: [asterisk-users] Weird stalling of playback on IAX2 channels on 1.8 svn I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Bryant, That's a relief, I thought it was just me ! Perhaps you can add something to https://bugs.digium.com/view.php?id=18110 T. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk -- _ -- 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] Polycom getting DCHP address from wrong VLAN
One more thing: Make sure that the port going to your data-DHCP server doesn't have the voice VLAN set on it. I troubleshot an installation for a few hours before thinking of this... Bests, Seb On 2010-10-08, at 2:37 AM, Thermal Wetland wrote: Hello, I have been tearing my hair out on this issue for 2 days, any help would be appreciated. We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch There are two VLANs, 1(data) 50(VoIP). When Polycoms are connected to the switch with VLAN 50 hard coded in the config they grab a DHCP address from VLAN 1, the PVID for the switch port. The ports have membership in VLAN 1 as the PVID and VLAN 50 as tagged traffic. I know the VoIP DHCP server is working because if I change a port to have a PVID of 50 any device gets the address from the VoIP DHCP server. I have tried the ports as 'general' and 'trunk' with no success. Any help would be greatly appreciated, I don't have much hair left! -- -Thermal -- _ -- 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] asterisk-users Digest, Vol 75, Issue 7
On Thu, Oct 7, 2010 at 11:25 PM, Fazil Amaan fazil_d...@yahoo.com wrote: Hi, I cannot get asterisk to start again after the g729 install failed. kindly advise what's the problem. Thank's -- _ -- 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 http://www.catb.org/esr/faqs/smart-questions.html#beprecise try starting asterisk with like asterisk -cv and see if it says anything of interest? -- _ -- 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] Polycom getting DCHP address from wrong VLAN
On Fri, Oct 8, 2010 at 5:16 AM, Sebastien Thomas li...@amplisys.ca wrote: One more thing: Make sure that the port going to your data-DHCP server doesn't have the voice VLAN set on it. I troubleshot an installation for a few hours before thinking of this... Interesting, the DHCP server for the voice and data are coming from the same router. The router connects to the switch via a trunk port. I will set up a dedicated DHCP server on a port with a PVID of 50. Thanks for the tip! -- _ -- 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] OpenVPN tunnel and one-way audio - Do I still need a SIP proxy?
Glad to hear it helped you Dennison. VPN is such a confusing beast to lots of people I think and hence the responses to this thread were all sort of work around and sometimes off-topic. It's also not well documented or maybe the feature is not widely used within the Asterisk community. I think it would be very good if some standard guidelines become available from the Asterisk side on this. Good day, Bruce On Wed, Oct 6, 2010 at 7:33 PM, Dennison Williams dennison.willi...@gmail.com wrote: On 09/22/2010 08:36 AM, Carlos Chavez wrote: Do you have a localnet statement in your sip.conf? That and using nat=no will make sure Asterisk does not replace the IP address in the Invite. I just wanted to give a +1 for this response. I am using openvpn to connect road warriors and remote offices to a central asterisk server. When setting up the configuration for the road warriors I created a new subnet for them, but forgot to include their subnet as a localnet directive in sip.conf. The result was that sip clients on the road warrior network would be able to register, but then when initiating a sip call the 200 response (to the INVITE from the client) from the asterisk server would include a contact address for some external ip that I did not recognize. This hint here allowed me to fix this bug, now calls from the road warrior subnet are coming in fine. Thanks! -- _ -- 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] SPA-2102 sending local IP instead of WAN IP in SIP packets
Kyle, Got an empty response from you. Were you intending to give your feedback? Regards, Bruce On Wed, Oct 6, 2010 at 8:10 PM, Kyle Kienapfel doctor.w...@gmail.comwrote: On Wed, Oct 6, 2010 at 12:50 PM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, This is such an annoying issue whenever it comes up. The sender and receive always receive the source public IP no matter what in the IP packets but then SIP packets go out with something like 192.168.0.20. In this instance, a Bell Canada DSL modem is installed and I saw the SPA-2102 register properly but only to fail later on due to sending it's LAN IP to the Asterisk server. So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also put the device on DMZ in the Bell Canada DSL modem and still the same issue. I am wondering when would manufacturers finally fully comply the SIP RFC?! I don't have the luxury to put SIP proxy, do a VPN, install a server or anything on client site. Diagram: Asterisk Server = Internet = Bell Canada Modem = SPA2102 Please send me your suggestions on how to fix this if you have this type of experience with SPA-2102 Thanks for the input, Bruce -- _ -- 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 Are you using stun? http://en.wikipedia.org/wiki/Session_Traversal_Utilities_for_NAT -- _ -- 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] SPA-2102 sending local IP instead of WAN IP in SIP packets
He sent a two liner On Fri, Oct 8, 2010 at 2:25 PM, bruce bruce bruceb...@gmail.com wrote: Kyle, Got an empty response from you. Were you intending to give your feedback? Regards, Bruce Are you using stun? http://en.wikipedia.org/wiki/Session_Traversal_Utilities_for_NAT -- _ -- 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] SPA-2102 sending local IP instead of WAN IP in SIP packets
On 10/06/2010 02:50 PM, bruce bruce wrote: Hi Guys, This is such an annoying issue whenever it comes up. The sender and receive always receive the source public IP no matter what in the IP packets but then SIP packets go out with something like 192.168.0.20. In this instance, a Bell Canada DSL modem is installed and I saw the SPA-2102 register properly but only to fail later on due to sending it's LAN IP to the Asterisk server. So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also put the device on DMZ in the Bell Canada DSL modem and still the same issue. I am wondering when would manufacturers finally fully comply the SIP RFC?! Exactly how is this behavior non-compliant with the (sic) SIP RFC? There is nothing in any SIP RFC that mandates that a SIP UA must be aware of multiple IP addresses over which it can be reached, and select the proper one to include in SIP requests and responses. In fact, many SIP UAs, Asterisk included, work just fine behind NAT devices without ever knowing what their external IP addresses are. If you had actually described how the device failed, we might be able to tell you what you could do to resolve the problem. In general, Asterisk works just fine with endpoints that are behind NAT devices and never send their external IP addresses in their SIP messages... there are probably millions of devices working that way every day. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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] looking for a better ATA
I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF talkoff and T38 Any advise before I start testing with these brands would be apperciated. Any better option you may know of. Thanks for any input Bryant -- _ -- 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] looking for a better ATA
On Fri, 8 Oct 2010, Bryant Zimmerman wrote: I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF talkoff and T38 Any advise before I start testing with these brands would be apperciated. Any better option you may know of. Thanks for any input Bryant I'm curious which of the above ills you attribute to the Linksys (assuming an SPA2102? The PAP2T does have the T38 problem I believe). Its basically the defacto standard for all the giant ITSPs. Perhaps your problem is one that could be rectified in some way. I have also tried Grandstream and Audiocodes (still use the MP-124s in certain situations) and have found that the SPA2102s work the best for us... 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] SIP NOTIFY to make linksys/cisco SPA BLF go yellow
Hi, I was wondering if anyone stumbled upon the correct event in a sip NOTIFY (from a SUBSCRIBE) to make the BLF lamps on a Linksys/Cisco SPA9xx/5xx go yellow? I'm trying to differentiate between On the Phone and DND with the BLF. Thanks. -- James -- _ -- 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] looking for a better ATA
Us too. Tons of SPA2102's out there working fine! On Fri, Oct 8, 2010 at 4:36 PM, Jeff LaCoursiere j...@sunfone.com wrote: On Fri, 8 Oct 2010, Bryant Zimmerman wrote: I currently us Linksys/Ciscio, Grandstream and AudioCodes ata's. none of the three perform well in all enviroments. Between stablity issues, T38 and DTMF talkoff all three suffer some combination of issues. I am looking at Patton and Innomedia. Has any one tried either brand and what is your experience with them. Which would be the base for stability, audio quality, provisioning, DTMF talkoff and T38 Any advise before I start testing with these brands would be apperciated. Any better option you may know of. Thanks for any input Bryant I'm curious which of the above ills you attribute to the Linksys (assuming an SPA2102? The PAP2T does have the T38 problem I believe). Its basically the defacto standard for all the giant ITSPs. Perhaps your problem is one that could be rectified in some way. I have also tried Grandstream and Audiocodes (still use the MP-124s in certain situations) and have found that the SPA2102s work the best for us... 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 -- _ -- 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] SPA-2102 sending local IP instead of WAN IP in SIP packets
Thanks for the feedback. I said previously, Asterisk receives packets like extens...@192.168.0.10 is trying to register to it. So, Asterisk sends out to local LAN an ACK which obviously is not right. SPA-2102 should send SIP request like extens...@123.123.123.123 (public IP). Thanks On Fri, Oct 8, 2010 at 3:32 PM, Kevin P. Fleming kpflem...@digium.comwrote: On 10/06/2010 02:50 PM, bruce bruce wrote: Hi Guys, This is such an annoying issue whenever it comes up. The sender and receive always receive the source public IP no matter what in the IP packets but then SIP packets go out with something like 192.168.0.20. In this instance, a Bell Canada DSL modem is installed and I saw the SPA-2102 register properly but only to fail later on due to sending it's LAN IP to the Asterisk server. So, I put NAT=yes and NAT_ALIVE=yes but that didn't help at all. I also put the device on DMZ in the Bell Canada DSL modem and still the same issue. I am wondering when would manufacturers finally fully comply the SIP RFC?! Exactly how is this behavior non-compliant with the (sic) SIP RFC? There is nothing in any SIP RFC that mandates that a SIP UA must be aware of multiple IP addresses over which it can be reached, and select the proper one to include in SIP requests and responses. In fact, many SIP UAs, Asterisk included, work just fine behind NAT devices without ever knowing what their external IP addresses are. If you had actually described how the device failed, we might be able to tell you what you could do to resolve the problem. In general, Asterisk works just fine with endpoints that are behind NAT devices and never send their external IP addresses in their SIP messages... there are probably millions of devices working that way every day. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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 -- _ -- 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
