Re: [asterisk-users] Trunk issue
Are you using freeswitch, or just plain asterisk? I just setup a trunk between Asterisk and CM this morning, and it works great providing that you allow for anonymous calls. -Original Message- From: Haley,Scott A scott.ha...@edwardjones.com Sent: Wednesday, April 23, 2014 9:36am To: asterisk-users@lists.digium.com asterisk-users@lists.digium.com Subject: [asterisk-users] Trunk issue -- _ -- 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-usersI have setup a trunk on Asterisk 11.7 to an Avaya Session Manager. Every time I try to send a call over it, the call gets rejected. Here is the sip debug trace. Could anyone tell me what may be going wrong? nxdasterisk-2*CLI [Apr 23 08:20:59] WARNING[19047]: pbx_spool.c:309 safe_append: Unable to set utime on /var/spool/asterisk/outgoing/scott.call: Operation not permitted Audio is at 18380 Adding codec 14 (alaw) to SDP Adding codec 100012 (g722) to SDP Adding codec 13 (ulaw) to SDP Reliably Transmitting (no NAT) to 192.168.175.135:5060: INVITE sip:913145152244@192.168.175.135 SIP/2.0 Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632 Max-Forwards: 70 From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f To: sip:913145152244@192.168.175.135 Contact: sip:3145152000@192.168.122.57:5060 Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.7.0 Date: Wed, 23 Apr 2014 13:20:59 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 229 v=0 o=root 424150695 424150695 IN IP4 192.168.122.57 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.122.57 t=0 0 m=audio 18380 RTP/AVP 8 9 0 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- --- SIP read from UDP:192.168.175.135:5060 --- SIP/2.0 100 Trying Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 CSeq: 102 INVITE From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f To: sip:913145152244@192.168.175.135 Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632 Content-Length: 0 - --- (7 headers 0 lines) --- --- SIP read from UDP:192.168.175.135:5060 --- INVITE sip:913145152...@devjones.com SIP/2.0 P-AV-Message-Id: 1_1 Route: sip:192.168.122.57;lr;phase=terminating Supported: replaces, timer Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Date: Wed, 23 Apr 2014 13:20:59 GMT Contact: sip:3145152000@192.168.122.57:5060;gsid=d13ae820-caef-11e3-9b9c-6c3be5a59e68 Via: SIP/2.0/UDP 192.168.175.135;rport;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4 Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967 Via: SIP/2.0/UDP 192.168.175.130:15060;rport;ibmsid=local.1389145532068_1778704_1816625;branch=z9hG4bK605195140054947 Via: SIP/2.0/UDP 192.168.175.135;rport=5060;branch=z9hG4bK6add1632-AP;ft=192.168.175.135~13c4;received=192.168.175.135 Via: SIP/2.0/UDP 192.168.122.57:5060;branch=z9hG4bK6add1632 Record-Route: sip:2ca13a6d@192.168.175.135;transport=udp;lr Record-Route: sip:192.168.175.130:15060;transport=udp;ibmsid=local.1389145532068_1778704_1816625;lr Record-Route: sip:2ca13a6d@192.168.175.135;transport=udp;lr P-Charging-Vector: icid-value=d13ae820-caef-11e3-9b9c-6c3be5a59e68 User-Agent: Asterisk PBX 11.7.0 AVAYA-SM-6.3.1.0.631004 P-Asserted-Identity: Edward Jones sip:3145152...@devjones.com From: Edward Jones sip:3145152000@192.168.122.57;tag=as4eecf94f To: sip:913145152244@192.168.175.135 Call-ID: 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 Max-Forwards: 66 CSeq: 102 INVITE Content-Type: application/sdp Content-Length: 229 Av-Global-Session-ID: d13ae820-caef-11e3-9b9c-6c3be5a59e68 P-Location: SM;origlocname=Asterisk-2;origsiglocname=Asterisk-2;origmedialocname=Asterisk-2;termlocname=Asterisk-2;termsiglocname=Asterisk-2;smaccounting=true v=0 o=root 424150695 424150695 IN IP4 192.168.122.57 s=Asterisk PBX 11.7.0 c=IN IP4 192.168.122.57 t=0 0 m=audio 18380 RTP/AVP 8 9 0 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv - --- (27 headers 11 lines) --- Sending to 192.168.175.135:5060 (no NAT) Sending to 192.168.175.135:5060 (no NAT) Using INVITE request as basis request - 504b8ce74a81e0f90ba457e77e8c9e60@192.168.122.57:5060 Found peer 'SMtrunk' for '3145152000' from 192.168.175.135:5060 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 0 Found audio description format PCMA for ID 8 Found audio description format G722 for ID 9 Found audio description
[asterisk-users] Asterisk as a media gateway
I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting. This is what I was thinking of trying. 1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc). 2. The mains server will not handle RTP traffic, it will send the RTP traffic to another system (another asterisk box?) for processing. At the end of the day, what I am hoping for is to have 1 brain, and mutiple work horse audio gateways that can be added and removed as needed. Has this been done? Can anyone point me to some documentation on how others have done this? It's always fun to play Richard Seguin. -- _ -- 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 QOS
I asked this on the list over the weekend, and likely missed a few people
inboxes.
I'm having a problem pulling data from RTPAUDIOQOS.For testing purposes I
have asterisk sending QOS data to the console. It seems I get QOS data only
if the caller hangs up, with the variable being empty if the callee (or
asterisk) hangs up.
Any idea why I would see this?
exten = h,1,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS})
Thanks,
Richard Seguin--
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[asterisk-users] RTPAUDIOQOS - Depending on who hangs up the phone, it's empty
I'm having a problem pulling data from RTPAUDIOQOS.For testing purposes I
have asterisk sending QOS data to the console. It seems I get QOS data only
if the caller hangs up, with the variable being empty if the callee (or
asterisk) hangs up.
Any idea why I would see this?
exten = h,1,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS})
Thanks,
Richard Seguin
--
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[asterisk-users] Access PBX from internet - best practice
Hello, I have a question about best practice (or recommended practice) for allowing SIP registrations from the Internet. This is what I was thinking of implementing: 1. Use OpenSips for the SBC, enable SRTP and TLS 2. Allow limited access to the actual Asterisk PBX (behind firewall) via OpenSips Is there anything that I am missing that probably should be implemented? Thanks, Richard -- _ -- 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] Access PBX from internet - best practice
The endpoints do not have a fixed IP, and a VPN tunnel wouldn't work under this scenario. Basically this setup is for people who are traveling, and may be using a smart phone at an airport (or something similar). The idea is that our system can be used to reduce toll costs, and provide access to internal resources. Thank you for the recommendations on fail2ban, IPtables, and the device naming scheme... I am not overly found of having a device name (ex: 101) that corresponds to the extension being used, so I will be using user and devices under freebpbx to name them differently. -Original Message- From: Administrator TOOTAI ad...@tootai.net Sent: Thursday, October 17, 2013 6:56am To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Access PBX from internet - best practice Le 17/10/2013 12:30, richard.seg...@marisec.ca a écrit : Hello, Hello I have a question about best practice (or recommended practice) for allowing SIP registrations from the Internet. Registrations from Internet is vague: - are EP with fixed IP: define the extension in SIP.conf with host = EP IP. You can even add an iptables rule to allow the EP IP to connect to port 5060 in udp (if your setup is this one) - are EP travellers = fail2ban or through VPN. OpenVPN is a good solution. This is what I was thinking of implementing: 1. Use OpenSips for the SBC, enable SRTP and TLS All clients doesn't support SRTP 2. Allow limited access to the actual Asterisk PBX (behind firewall) via OpenSips Is there anything that I am missing that probably should be implemented? In all cases I would recommend: - a strong extension definition eg [MyFav0Rite-prefiX_123] instead of [123] - always use fail2ban [...] -- Daniel -- _ -- 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
