[asterisk-users] SIP Jitterbuffer
Hello people What are the cons, if any, of enabling a jitterbuffer? We are currently using version 1.8 Thanks in advance Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] SIP Jitterbuffer
> What are the cons, if any, of enabling a jitterbuffer? Memory and latency. -- _ -- 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] ports, routers and firewalls
I just want to make a SIP call from 192.168.1.3 to 192.168.1.4; or not even a call. Ring? Beep? Ping? Some sort of "hello world" connection. 192.168.1.1 netgear router 192.168.1.2 asterisk (vicidial) 192.168.1.3 ubuntu client 192.168.1.4 mac OSX client (not shown) Do I have a firewall problem which would impact a soft phone from establishing a connection? thufir@doge:~$ thufir@doge:~$ thufir@doge:~$ nmap 192.168.1.1 Starting Nmap 6.46 ( http://nmap.org ) at 2015-02-18 06:10 PST Nmap scan report for 192.168.1.1 Host is up (0.0086s latency). Not shown: 994 closed ports PORT STATE SERVICE 23/tcpopen telnet 53/tcpopen domain 80/tcpopen http /tcp open dec-notes /tcp open freeciv 49152/tcp open unknown Nmap done: 1 IP address (1 host up) scanned in 0.14 seconds thufir@doge:~$ thufir@doge:~$ nmap 192.168.1.2 Starting Nmap 6.46 ( http://nmap.org ) at 2015-02-18 06:10 PST Nmap scan report for 192.168.1.2 Host is up (0.00027s latency). Not shown: 997 filtered ports PORTSTATE SERVICE 22/tcp open ssh 80/tcp open http 443/tcp open https Nmap done: 1 IP address (1 host up) scanned in 4.95 seconds thufir@doge:~$ thufir@doge:~$ thufir@doge:~$ ssh thufir@192.168.1.2 Password: Last login: Mon Feb 16 00:43:01 2015 from 192.168.1.2 Thank you for installing ViciBox Server v.6.0! This software is available for free download at http://www.vicibox.com. If you paid for this software you have been ripped off. Please report any fraud or abuses of this software to ab...@vicidial.com. Please report any bugs on the forum at http://www.vicidial.org To configure the LAN settings type: yast lan To change the server IP in the database type: /usr/share/astguiclient/ADMIN_update_server_ip.pl Official paid-for ViciDial support is available at http://www.vicidial.com Free community-based ViciDial Support is available at http://www.vicidial.org/VICIDIALforum - ViciBox Redux v.6.0.3-141118 Could not chdir to home directory /home/thufir: No such file or directory thufir@tleilax:/> thufir@tleilax:/> nmap 192.168.1.3 Starting Nmap 6.40 ( http://nmap.org ) at 2015-02-18 09:14 EST Nmap scan report for 192.168.1.3 Host is up (0.00075s latency). Not shown: 998 closed ports PORT STATE SERVICE 22/tcp open ssh 2000/tcp open cisco-sccp Nmap done: 1 IP address (1 host up) scanned in 0.15 seconds thufir@tleilax:/> thufir@tleilax:/> thanks, Thufir -- _ -- 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] Respond with 200 OK on OPTIONS
On Tue, 17 Feb 2015 08:28:31 -0600, Matthew Jordan wrote: > Asterisk attempts to look up who the OPTIONS request is for, using the > username portion of the request URI. Make sure you have a matching > extension for what your upstream provider is sending you, and chan_sip > will respond with a 200 OK. In general, this 200 OK status code can be used for troubleshooting? Is there a log of status codes sent, or that's just done live through the console? thanks, Thufir -- _ -- 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 13 - sorcery realtime for pjsip publish objects
Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 (Internal: ext. 10015) -- _ -- 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 13 - sorcery realtime for pjsip publish objects
Matt Hoskins wrote: Hello, Kia ora, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. It "should" be possible with recent changes that have been done. I personally have not tried it, and it would still require a reload to pick up changes regardless. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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] SIP trunk no audio
I have two machines on the internet. Box A and Box B. Box A has a SIP trunk to the world, Making calls box A works fine I have audio to my cell and all works. I defined a SIP trunk between box B and A. tried to make a call originating from box B - to box A and then over the SIP trunk to my cell. My cell rings but then no audio. I have defined SIP trunks before between boxes pretty straight forward. I have checked and my firewalls are open for SIP/RTP -A INPUT -m state --state NEW -m udp -p udp --dport 5060 -j ACCEPT -A INPUT -m state --state NEW -m tcp -p tcp --dport 8000:6 -j ACCEPT -A INPUT -m state --state NEW -m udp -p udp --dport 8000:6 -j ACCEPT I am using asterisk 11.16 box A is [boxab_sip] type=friend username=boxa_sip secret=*** disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 host=DNS Name here context=sip_trunk insecure=port,invite box B is [boxab_sip] type=friend username=boxab_sip secret=*** disallow=all allow=ulaw allow=alaw allow=gsm dtmfmode=rfc2833 host=DNS Name here context=sip_turnk insecure=port,invite Is there something I am missing? The one piece I have not done before is SIP trunk - to - SIP trunk. But the phone rings - so its routed - just no audio. Thoughts? 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] Res_fax - FAXOPT(faxdetect)
Hello
Le 17/02/2015 17:00, Administrator TOOTAI a écrit :
Hi,
as stated in the documentation, it's allowed to set
FAXOPT(faxdetect)=yes/no to allow fax detection.
It's done (see below) but still fax detection :-( Extension 300 is
hylafax with iaxmodem.
On the upper Asterisk gw it's the same, despite the faxdetect set to no
we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile
phone calling the 0123456789 PSTN number.
-- Executing [0123456789@from-internal:1]
Set("SIP/TOOTAi-8262", "FAXOPT(faxdetect)=no") in new stack
-- Executing [0123456789@from-internal:2]
Macro("SIP/TOOTAi-8262", "Fax") in new stack
-- Executing [s@macro-Fax:1] Dial("SIP/TOOTAi-8262",
"IAX2/300,,") in new stack
-- Called IAX2/300
-- Call accepted by 127.0.0.1 (format alaw)
-- Format for call is (alaw)
-- IAX2/300-7211 is ringing
-- IAX2/300-7211 answered SIP/TOOTAiAudio-8262
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
[2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645
process_sdp: T.38 re-INVITE detected but no fax extension
[2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868
process_sdp: Insufficient information for SDP (m= not found)
-- Executing [h@from-internal:1] Hangup("SIP/TOOTAi-8262", "")
in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/TOOTAi-8262'
-- Hungup 'IAX2/300-7211'
Thanks for your support
No one have an idea on this ?
--
Daniel
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Re: [asterisk-users] Res_fax - FAXOPT(faxdetect)
I solved the issue by not answering the call as I assume others have done.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Wednesday, February 18, 2015 12:50 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Res_fax - FAXOPT(faxdetect)
Hello
Le 17/02/2015 17:00, Administrator TOOTAI a écrit :
> Hi,
>
> as stated in the documentation, it's allowed to set
> FAXOPT(faxdetect)=yes/no to allow fax detection.
>
> It's done (see below) but still fax detection :-( Extension 300 is
> hylafax with iaxmodem.
>
> On the upper Asterisk gw it's the same, despite the faxdetect set to no
> we also have the NOTICE of T.38 re-INVITE. Test is done with a mobile
> phone calling the 0123456789 PSTN number.
>
> -- Executing [0123456789@from-internal:1]
> Set("SIP/TOOTAi-8262", "FAXOPT(faxdetect)=no") in new stack
> -- Executing [0123456789@from-internal:2]
> Macro("SIP/TOOTAi-8262", "Fax") in new stack
> -- Executing [s@macro-Fax:1] Dial("SIP/TOOTAi-8262",
> "IAX2/300,,") in new stack
> -- Called IAX2/300
> -- Call accepted by 127.0.0.1 (format alaw)
> -- Format for call is (alaw)
> -- IAX2/300-7211 is ringing
> -- IAX2/300-7211 answered SIP/TOOTAiAudio-8262
>== Using UDPTL TOS bits 184
>== Using UDPTL CoS mark 5
> [2015-02-17 16:52:51] NOTICE[3467][C-1d5b]: chan_sip.c:10645
> process_sdp: T.38 re-INVITE detected but no fax extension
> [2015-02-17 16:52:56] WARNING[3467][C-1d5b]: chan_sip.c:9868
> process_sdp: Insufficient information for SDP (m= not found)
> -- Executing [h@from-internal:1] Hangup("SIP/TOOTAi-8262", "")
> in new stack
>== Spawn extension (from-internal, h, 1) exited non-zero on
> 'SIP/TOOTAi-8262'
> -- Hungup 'IAX2/300-7211'
>
> Thanks for your support
>
No one have an idea on this ?
--
Daniel
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Re: [asterisk-users] SIP trunk no audio
But the phone rings - so its routed - just no audio. The ringing is SIP signaling. The audio is RTP data. See if the audio is getting routed with a sniffer. Maybe use one codec that both clients support. Adrian Serafini -- _ -- 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 13 - sorcery realtime for pjsip publish objects
Excellent. I was using ast-13.1.0 with no luck. I upgraded to 13.2.0 and have made it further, but am having a little difficulty. The outbound-publish object types seems to be working in realtime now. But the asterisk-publication object is only reading from sorcery.conf. I know you said that it *should* work, with no guarantee, which I'm fine with. I just want to make sure I don't have a possible misconfiguration issue. Here is my sorcery.conf and extconfig file: Sorcery.conf [res_pjsip] endpoint=realtime,ps_endpoints auth=realtime,ps_auths aor=realtime,ps_aors domain_alias=realtime,ps_domain_aliases [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips [res_pjsip_outbound_publish] outbound-publish=realtime,ps_outbound_publish [res_pjsip_pubsub] inbound-publication=realtime,ps_inbound_publication [res_pjsip_publish_asterisk] asterisk-publication=realtime,ps_asterisk_publication extconfig.conf --- ps_endpoints => odbc,asterisk-realtime ps_auths => odbc,asterisk-realtime ps_aors => odbc,asterisk-realtime ps_domain_aliases => odbc,asterisk-realtime ps_endpoint_id_ips => odbc,asterisk-realtime ps_outbound_publish => odbc,asterisk-realtime ps_inbound_publication => odbc,asterisk-realtime ps_asterisk_publication => odbc,asterisk-realtime Matt Hoskins | NPG Corp | Systems Architect 816.749.2815 (Internal: ext. 10015) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Wednesday, February 18, 2015 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk 13 - sorcery realtime for pjsip publish objects Matt Hoskins wrote: > Hello, Kia ora, > I am currently trying to set up pjsip realtime and would like to have > outbound-publish, inbound-publication, and asterisk-publication > sorcery object types in ODBC realtime. Is that currently supported? I > know that some object types are known working and others are not. I > was curious what the status of those objects are. It "should" be possible with recent changes that have been done. I personally have not tried it, and it would still require a reload to pick up changes regardless. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5kaWdpdW0uY2 9t&_r=YmFzZQ%3D%3D & http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5hc3Rlcmlzay 5vcmc%3D&_r=YmFzZQ%3D%3D -- _ -- Bandwidth and Colocation Provided by http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5hcGktZGlnaX RhbC5jb20%3D&_r=YmFzZQ%3D%3D -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL3d3dy5hc3Rlcmlzay 5vcmcvaGVsbG8%3D&_r=YmFzZQ%3D%3D asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://spamaway.npgco.com/canit/urlproxy.php?_q=aHR0cDovL2xpc3RzLmRpZ2l1bS 5jb20vbWFpbG1hbi9saXN0aW5mby9hc3Rlcmlzay11c2Vycw%3D%3D&_r=YmFzZQ%3D%3D -- BEGIN-ANTISPAM-VOTING-LINKS -- Teach CanIt if this mail (ID 01NRQTiBC) is spam: Spam: http://spamaway.npgco.com/canit/b.php?i=01NRQTiBC&m=355a208e3378&t=2015021 8&c=s Not spam: http://spamaway.npgco.com/canit/b.php?i=01NRQTiBC&m=355a208e3378&t=2015021 8&c=n Forget vote: http://spamaway.npgco.com/canit/b.php?i=01NRQTiBC&m=355a208e3378&t=2015021 8&c=f -- END-ANTISPAM-VOTING-LINKS -- _ -- 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] Callfile problem - Unable to find codec translation path from (nothing)
Joshua, If I'm understanding this correctly, you're saying that the Playback is failing because it isn't connected to anything on the other end, because the Dial() failed. When the channel is created on the "OutgoingSpoolFailed" extension, what context is it created in, one of the origin legs? Is there a way detect this condition in the target context ([outbound-swift]), or better yet, verify the other leg is attached before starting the logic? -Justin -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp Sent: Tuesday, February 17, 2015 4:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Callfile problem - Unable to find codec translation path from (nothing) Justin Killen wrote: > > Whenever I try to copy this callfile into > /var/spool/asterisk/outgoing/ I get these 3 lines repeating over and > over (I'm not 100% sure which entry is first): > > [2015-02-16 16:56:02] WARNING[9737][C-f8a7]: channel.c:5353 > set_format: Unable to find a codec translation path from (nothing) to > (slin) > > [2015-02-16 16:56:02] WARNING[9737][C-f8a7]: file.c:1017 > ast_streamfile: Unable to open AAA/check_ip_failure (format (nothing)): > Function not implemented > > [2015-02-16 16:56:02] WARNING[9737][C-f8a7]: app_playback.c:484 > playback_exec: ast_streamfile failed on OutgoingSpoolFailed for > AAA/check_ip_failure > > [2015-02-16 16:56:02] WARNING[9737][C-f8a7]: channel.c:5353 > set_format: Unable to find a codec translation path from (nothing) to > (slin) > > [2015-02-16 16:56:02] WARNING[9737][C-f8a7]: file.c:1017 > ast_streamfile: Unable to open AAA/check_ip_failure (format (nothing)): > Function not implemented > > [2015-02-16 16:56:02] WARNING[9737][C-f8a7]: app_playback.c:484 > playback_exec: ast_streamfile failed on OutgoingSpoolFailed for > AAA/check_ip_failure > > Is there something special I need to do to trick the translation into > doing the right thing? It can never do the right thing there. If the origination fails for some reason then a channel (without any formats) is created to the "OutgoingSpoolFailed" extension. Due to the way you've written your dialplan logic this will attempt to do things with media. Since it's not a real channel and has no formats, that will fail. Since your dialplan logic also has it go in a loop it just goes 'round and 'round. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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] Callfile problem - Unable to find codec translation path from (nothing)
Justin Killen wrote: Joshua, If I'm understanding this correctly, you're saying that the Playback is failing because it isn't connected to anything on the other end, because the Dial() failed. When the channel is created on the "OutgoingSpoolFailed" extension, what context is it created in, one of the origin legs? Is there a way detect this condition in the target context ([outbound-swift]), or better yet, verify the other leg is attached before starting the logic? It is created in the context you have told the answered channel to go to. One way to fix it would be to add an OutgoingSpoolFailed extension for each priority so that Swift isn't invoked. You could also use GotoIf in the first priority and go elsewhere if the extension is OutgoingSpoolFailed. There exist many options. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US 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] sipsak: 404 error
Hi, I **think** that I have user of thufir101, because I get a 200 response below, but I also get a 404. It seems to depend on how I send the ip address/fqdn? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201password 201 default No Yes thufir101 password thufir101 default No Yes babyteljlfjd54545 default No Yes gs102 X58sKpZCcDfcGT0 gs102 default No Yes tleilax*CLI> tleilax*CLI> sip show user thufir101 * Name : thufir101 Secret : MD5Secret: Context : default Language : en Accountcode : thufir101 AMA flags: Unknown Netborder CPD: No Transfer mode: open MaxCallBR: 384 kbps CallingPres : Presentation Allowed, Not Screened Call limit : 0 Callgroup: Pickupgroup : Callerid : "atreides" <123> ACL : No Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Sess-Min-SE : 90 secs RTP Engine : asterisk Codec Order : (ulaw:20,gsm:20) Auto-Framing: No tleilax*CLI> which would make the URI sip:thufir...@tleilax.bounceme.net ? thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:thufir101@tleilax No SRV record: _sip._tcp.tleilax No SRV record: _sip._udp.tleilax using A record: tleilax message received: SIP/2.0 200 OK CSeq: 1 OPTIONS Via: SIP/2.0/UDP 127.0.1.1:52173;branch=z9hG4bK.4ca3965f;rport=52173;alias;received=192.168.1.3 User-Agent: Ekiga/4.0.1 From: sip:sipsak@127.0.1.1:52173;tag=631bb564 Call-ID: 1662760292@127.0.1.1 To: sip:thufir101@tleilax Contact: Content-Length: 0 ** reply received after 3.381 ms ** SIP/2.0 200 OK final received thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:thufir...@tleilax.bounceme.net No SRV record: _sip._tcp.tleilax.bounceme.net No SRV record: _sip._udp.tleilax.bounceme.net using A record: tleilax.bounceme.net message received: SIP/2.0 200 OK CSeq: 1 OPTIONS Via: SIP/2.0/UDP 127.0.1.1:35077;branch=z9hG4bK.353a619c;rport=35077;alias;received=192.168.1.3 User-Agent: Ekiga/4.0.1 From: sip:sipsak@127.0.1.1:35077;tag=239b4596 Call-ID: 597378454@127.0.1.1 To: sip:thufir...@tleilax.bounceme.net Contact: Content-Length: 0 ** reply received after 2.987 ms ** SIP/2.0 200 OK final received thufir@doge:~$ thufir@doge:~$ sudo sipsak -vv -s sip:thufir101@192.168.1.2 message received: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.1.1:39721;branch=z9hG4bK.4b7b7fab;alias;received=192.168.1.3;rport=39721 From: sip:sipsak@127.0.1.1:39721;tag=6b70e831 To: sip:thufir101@192.168.1.2;tag=as34aa76ca Call-ID: 1802561585@127.0.1.1 CSeq: 1 OPTIONS Server: Asterisk PBX 1.8.32.1-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 ** reply received after 0.665 ms ** SIP/2.0 404 Not Found final received thufir@doge:~$ thufir@doge:~$ I updated my hosts file on doge with the ip adress for tleilax...for some reason that "makes" it work..? any pointers, thank you, Thufir -- _ -- 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] TimerFD errors if MTU size is set incorrectly - SIP trunk
Hi all Is there a relation between the above? I'm having a problem where I suspect my internet access provider (through whom I go to a SIP trunk provider) have got MTU size problems. My asterisk (1.8.11.0) is constantly going into the situation where a TimerFD error is spammed in the CLI, load goes up and up until the system is completely unusable. I have an admission by the ISP that their MTU size on their fiber NTU "may" be incorrect. Can an incorrect MTU size on an ethernet connection cause Asterisk to experience errors with the kernel timing source? Thank you Stefan -- _ -- 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
