Re: [asterisk-users] Open Source Asterisk Polling Solution
On 4/23/2014 12:20 AM, Nick Cameo wrote: That's about as simple as it gets. A call file that goes to the dialplan. A dialplan that consists of Read (which would play the message) followed a GotoIf into a mailbox (either voicemail or Dial() to an external number). One hint for doing unattended dialing like this, make sure you're dialing using a SIP or other digital method rather than, say, out an analogue port that doesn't have decent answer detect. And you can't just dump a whole bunch of call files into the system at once, you'll need to meter them out based on the number of concurrent outbound calls your provider will allow. Hello James, Good to see you here, and thank you very much. Though my basic idea of how it will look using call files and dialplan is like what you and others on here have pointed out. Yes, we are using SIP for both origination and termination (just helping my friend use some of our accounts used for PBX, and prepaid). I have been using * for many years now however, never for call center/predictive dialer type processes. Once I have got this thing to call out and get calls coming in. It would be nice to write to a database all the users that press option on. I have a strong Java, PHP and SQL background. Will probably need to make a call using AGI or such? N. You can go AGI, but there are direct ODBC handles available in the dialplan if you build Asterisk properly with the ODBC resources enabled. That'd my personal preference from a performance standpoint. -- _ -- 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] Open Source Asterisk Polling Solution
I agree that ODBC is the way to go here. It's trivially easy to setup, and equally simple to push database updates via the dialplan. I've used ODBC connectivity with Asterisk in a large and VERY busy call center, and performance was never remotely an issue (call recording is a different story, but that's something else entirely...). There was mention of checking against a DNC list, and ODBC would be good for this as well - just put that into a table and match against it before making your outbound call. -Josh On Wed, Apr 23, 2014 at 4:12 AM, James Sharp ja...@fivecats.org wrote: On 4/23/2014 12:20 AM, Nick Cameo wrote: That's about as simple as it gets. A call file that goes to the dialplan. A dialplan that consists of Read (which would play the message) followed a GotoIf into a mailbox (either voicemail or Dial() to an external number). One hint for doing unattended dialing like this, make sure you're dialing using a SIP or other digital method rather than, say, out an analogue port that doesn't have decent answer detect. And you can't just dump a whole bunch of call files into the system at once, you'll need to meter them out based on the number of concurrent outbound calls your provider will allow. Hello James, Good to see you here, and thank you very much. Though my basic idea of how it will look using call files and dialplan is like what you and others on here have pointed out. Yes, we are using SIP for both origination and termination (just helping my friend use some of our accounts used for PBX, and prepaid). I have been using * for many years now however, never for call center/predictive dialer type processes. Once I have got this thing to call out and get calls coming in. It would be nice to write to a database all the users that press option on. I have a strong Java, PHP and SQL background. Will probably need to make a call using AGI or such? N. You can go AGI, but there are direct ODBC handles available in the dialplan if you build Asterisk properly with the ODBC resources enabled. That'd my personal preference from a performance standpoint. -- _ -- 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] ICE
Gholamreza Sabery wrote: Hello, Kia ora, I have an Asterisk server with a public IP address and a bunch of clients. Most of my clients are behind NATs (sometimes two clients are behind the same NAT i.e in the same private network). I want to use ICE so that the clients behind the same NAT can send RTP traffic directly to each other and other clients use Asterisk or a TURN server. I tested a specific scenario using two Linphone 3.7 as clients behind the same NAT but finally traffic ended in Asterisk. I checked the packets; the first client sends its host and server reflexive candidates to Asterisk and Asterisk sends it's public address. Then Asterisk will send it's own address to second client and second client in the final 200 OK SIP only sends it's local address normally without using ICE (ICE is enabled on both clients). Using ICE for such a purpose is possible at all? Asterisk does not currently support this configuration (specifically passing through candidates and ICE information like this). 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] Trunk issue
I 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 format PCMU for ID 0 Capabilities: us - (ulaw|alaw|g722), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g722) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.122.57:18380 Looking for 913145152244 in from-pstn (domain devjones.com) --- Reliably Transmitting (no NAT) to 192.168.175.135:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.175.135;branch=z9hG4bK561174433949967-AP;ft=192.168.175.135~13c4;received=192.168.175.135;rport=5060 Via: SIP/2.0/UDP 192.168.175.130:15060;rport=15060;ibmsid=local.1389145532068_1778706_1816627;branch=z9hG4bK561174433949967 Via: SIP/2.0/UDP
Re: [asterisk-users] Trunk issue
Hello Le 23/04/2014 15:36, Haley,Scott A a écrit : I 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? [...] Here [Apr 23 08:20:59] NOTICE[19026][C-0003]: chan_sip.c:25450 handle_request_invite: Call from 'SMtrunk' (192.168.175.135:5060) to extension '913145152244' rejected because extension not found in context 'from-pstn'. [...] Regards -- 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
Re: [asterisk-users] Open Source Asterisk Polling Solution
On 4/23/2014 12:20 AM, Nick Cameo wrote: I have a strong Java, PHP and SQL background. Will probably need to make a call using AGI or such? On Wed, 23 Apr 2014, James Sharp wrote: You can go AGI, but there are direct ODBC handles available in the dialplan if you build Asterisk properly with the ODBC resources enabled. That'd my personal preference from a performance standpoint. On Wed, 23 Apr 2014, Josh Metzger wrote: I agree that ODBC is the way to go here. It's trivially easy to setup, and equally simple to push database updates via the dialplan. I've used ODBC connectivity with Asterisk in a large and VERY busy call center, and performance was never remotely an issue (call recording is a different story, but that's something else entirely...). There was mention of checking against a DNC list, and ODBC would be good for this as well - just put that into a table and match against it before making your outbound call. And in this corner... I always do database access it an AGI. IMNSHO, any significant chunk of logic or functionality belongs in an AGI. Keep your dialplan lean and mean. I tried database access in the dialplan using the mysql() application years ago, just to confirm I was right and I was :) What an ugly, messy, fragile dialplan. You already know database access in 'real' languages, why would you want to code in a limited and difficult to debug environment? I'm an 'old school' C programmer, so performance is always close to my heart, but not when it makes my job harder. Writing database access in the dialplan avoids creating a process for the AGI, but unless you're processing hundreds or thousands of calls per second, process creation is not going to be a factor. I write my AGIs in C. It is my 'sharpest tool in the toolbox.' If C is not in your 'wheelhouse,' use PHP or coughJava/cough. You (and the next guy who gets to enhance and maintain this application) will be glad you did. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Open Source Asterisk Polling Solution
I tried database access in the dialplan using the mysql() application years ago, just to confirm I was right and I was :) What an ugly, messy, fragile dialplan. With FuncODBC this is no longer an issue. All of the query logic is handled outside of the dial plan. Doug -- _ -- 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] Open Source Asterisk Polling Solution
I've always done my DB access via func_odbc and not with the mysql package. While we ran a MySQL db, I was more comfortable with the odbc stuff because it was part of Asterisk core and not an addon package. I can't speak to the simplicity of using the mysql stuff vs the odbc stuff, but there isn't a lot to creating your query in func_odbc and then calling it from your dialplan (and passing a few variables). I'd say it's no more difficult than calling an AGI, in my case we WERE processing a LARGE volume of calls, and importantly: It didn't require me to learn C or PHP. ;-) Now if you want complicated, somewhere I have a very long GotoIf() that includes an ODBC call and nested Math() functions... -Josh On Wed, Apr 23, 2014 at 11:17 AM, Doug Lytle supp...@drdos.info wrote: I tried database access in the dialplan using the mysql() application years ago, just to confirm I was right and I was :) What an ugly, messy, fragile dialplan. With FuncODBC this is no longer an issue. All of the query logic is handled outside of the dial plan. Doug -- _ -- 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] Asterisk 1.8.27.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.27.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 1.8.27.0 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:
Bugs fixed in this release:
---
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by Paolo Compagnini)
* ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
sip.conf.sample (Reported by Eugene)
* ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
minus signs (Reported by Jeremy Lainé)
* ASTERISK-23046 - Custom CDR fields set during a GoSUB called
from app_queue are not inserted (Reported by Denis Pantsyrev)
* ASTERISK-23027 - [patch] Spelling typo transfered instead of
transferred (Reported by Jeremy Lainé)
* ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
channel connects (Reported by Michael Cargile)
* ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
request and request queue may differ - fix for locking (Reported
by adomjan)
* ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
media offer due to invalid or unsupported syntax (Reported by
adomjan)
* ASTERISK-22861 - [patch]Specifying a null time as parameter to
GotoIfTime or ExecIfTime causes segmentation fault (Reported by
Sebastian Murray-Roberts)
* ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
exceeded (Reported by pz)
* ASTERISK-22662 - Documentation fix? - queues.conf says
persistentmembers defaults to yes, it appears to lie (Reported
by Rusty Newton)
* ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
handle selinux port restrictions (Reported by Corey Farrell)
* ASTERISK-23220 - STACK_PEEK function with no arguments causes
crash/core dump (Reported by James Sharp)
* ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
command multiple times on cli_aliases (Reported by Joel Vandal)
* ASTERISK-22757 - segfault in res_clialiases.so on reload when
mapping module reload command (Reported by Gareth Blades)
* ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
(Reported by LN)
* ASTERISK-23178 - devicestate.h: device state setting functions
are documented with the wrong return values (Reported by
Jonathan Rose)
* ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
res_parking.so is not loaded, or if res_parking.conf has no
configuration (Reported by CJ Oster)
* ASTERISK-23069 - Custom CDR variable not recorded when set in
macro called from app_queue (Reported by Bryan Anderson)
* ASTERISK-19499 - ConfBridge MOH is not working for transferee
after attended transfer (Reported by Timo Teräs)
* ASTERISK-23261 - [patch]Output mixup in
${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
* ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
variables for subsequent records (Reported by zvision)
* ASTERISK-23141 - Asterisk crashes on Dial(), in
pbx_find_extension at pbx.c (Reported by Maxim)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- probably introduced in 11.7.0 (Reported by OK)
* ASTERISK-23323 - [patch]chan_sip: missing p-owner checks in
handle_response_invite (Reported by Walter Doekes)
* ASTERISK-23382 - [patch]Build System: make -qp can corrupt
menuselect-tree and related files (Reported by Corey Farrell)
* ASTERISK-23406 - [patch]Fix typo in sip show peer (Reported by
ibercom)
* ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
(Reported by Jeremy Lainé)
* ASTERISK-23104 - Specifying the SetVar AMI without a Channel
cause Asterisk to crash (Reported by Joel Vandal)
* ASTERISK-23383 - Wrong sense test on stat return code causes
unchanged config check to break with include files. (Reported by
David Woolley)
* ASTERISK-17523 - Qualify for static realtime peers does not work
(Reported by Maciej Krajewski)
* ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
unload_module and do_monitor (Reported by Corey Farrell)
* ASTERISK-23373 - [patch]Security: Open FD exhaustion with
chan_sip Session-Timers (Reported by Corey Farrell)
* ASTERISK-23340 - Security Vulnerability: stack allocation of
cookie headers in loop allows for unauthenticated remote denial
of service attack (Reported by Matt Jordan)
* ASTERISK-23488 - Logic error in callerid checksum processing
(Reported by Russ Meyerriecks)
*
[asterisk-users] Asterisk 11.9.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 11.9.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 11.9.0 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:
Bugs fixed in this release:
---
* ASTERISK-22790 - check_modem_rate() may return incorrect rate
for V.27 (Reported by Paolo Compagnini)
* ASTERISK-23034 - [patch] manager Originate doesn't abort on
failed format_cap allocation (Reported by Corey Farrell)
* ASTERISK-23061 - [Patch] 'textsupport' setting not mentioned in
sip.conf.sample (Reported by Eugene)
* ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
minus signs (Reported by Jeremy Lainé)
* ASTERISK-23046 - Custom CDR fields set during a GoSUB called
from app_queue are not inserted (Reported by Denis Pantsyrev)
* ASTERISK-23027 - [patch] Spelling typo transfered instead of
transferred (Reported by Jeremy Lainé)
* ASTERISK-23008 - Local channels loose CALLERID name when DAHDI
channel connects (Reported by Michael Cargile)
* ASTERISK-23100 - [patch] In chan_mgcp the ident in transmitted
request and request queue may differ - fix for locking (Reported
by adomjan)
* ASTERISK-22988 - [patch]T38 , SIP 488 after Rejecting image
media offer due to invalid or unsupported syntax (Reported by
adomjan)
* ASTERISK-22861 - [patch]Specifying a null time as parameter to
GotoIfTime or ExecIfTime causes segmentation fault (Reported by
Sebastian Murray-Roberts)
* ASTERISK-17837 - extconfig.conf - Maximum Include level (1)
exceeded (Reported by pz)
* ASTERISK-22662 - Documentation fix? - queues.conf says
persistentmembers defaults to yes, it appears to lie (Reported
by Rusty Newton)
* ASTERISK-23134 - [patch] res_rtp_asterisk port selection cannot
handle selinux port restrictions (Reported by Corey Farrell)
* ASTERISK-23220 - STACK_PEEK function with no arguments causes
crash/core dump (Reported by James Sharp)
* ASTERISK-19773 - Asterisk crash on issuing Asterisk-CLI 'reload'
command multiple times on cli_aliases (Reported by Joel Vandal)
* ASTERISK-22757 - segfault in res_clialiases.so on reload when
mapping module reload command (Reported by Gareth Blades)
* ASTERISK-17727 - [patch] TLS doesn't get all certificate chain
(Reported by LN)
* ASTERISK-23178 - devicestate.h: device state setting functions
are documented with the wrong return values (Reported by
Jonathan Rose)
* ASTERISK-23232 - LocalBridge AMI Event LocalOptimization value
is opposite to what's expected (Reported by Leon Roy)
* ASTERISK-23098 - [patch]possible null pointer dereference in
format.c (Reported by Marcello Ceschia)
* ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
res_parking.so is not loaded, or if res_parking.conf has no
configuration (Reported by CJ Oster)
* ASTERISK-23069 - Custom CDR variable not recorded when set in
macro called from app_queue (Reported by Bryan Anderson)
* ASTERISK-19499 - ConfBridge MOH is not working for transferee
after attended transfer (Reported by Timo Teräs)
* ASTERISK-23261 - [patch]Output mixup in
${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
* ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic
payload change in rtp mapping in the 200 OK response (Reported
by NITESH BANSAL)
* ASTERISK-23255 - UUID included for Redhat, but missing for
Debian distros in install_prereq script (Reported by Rusty
Newton)
* ASTERISK-23260 - [patch]ForkCDR v option does not keep CDR
variables for subsequent records (Reported by zvision)
* ASTERISK-23141 - Asterisk crashes on Dial(), in
pbx_find_extension at pbx.c (Reported by Maxim)
* ASTERISK-23336 - Asterisk warning Don't know how to indicate
condition 33 on ooh323c on outgoing calls from H323 to SIP peer
(Reported by Alexander Semych)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- probably introduced in 11.7.0 (Reported by OK)
* ASTERISK-23323 - [patch]chan_sip: missing p-owner checks in
handle_response_invite (Reported by Walter Doekes)
* ASTERISK-23406 - [patch]Fix typo in sip show peer (Reported by
ibercom)
* ASTERISK-23310 - bridged channel crashes in bridge_p2p_rtp_write
(Reported by Jeremy Lainé)
* ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call
from hold (Reported by Vytis ValentinaviÄius)
* ASTERISK-23104 - Specifying the SetVar AMI without a Channel
cause Asterisk to crash (Reported by Joel
[asterisk-users] Asterisk 12.2.0 Now Available
The Asterisk Development Team has announced the release of Asterisk 12.2.0.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 12.2.0 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:
New Features made in this release:
---
* ASTERISK-23276 - Look at adding the 'pjsip show channel' command
(Reported by George Joseph)
Bugs fixed in this release:
---
* ASTERISK-23290 - chan_sip: ast_bridge_transfer_blind causes
channel to be hung up immediately, leading to BYE request being
sent before NOTIFY (Reported by Matt Jordan)
* ASTERISK-23098 - [patch]possible null pointer dereference in
format.c (Reported by Marcello Ceschia)
* ASTERISK-23125 - ARI: URI is case sensitive (Reported by Zane
Conkle)
* ASTERISK-23297 - Asterisk 12, pbx_config.so segfaults if
res_parking.so is not loaded, or if res_parking.conf has no
configuration (Reported by CJ Oster)
* ASTERISK-22738 - Security denial error in calls from H323
trunk (ooh323.c) (Reported by Gabriele Odone)
* ASTERISK-23069 - Custom CDR variable not recorded when set in
macro called from app_queue (Reported by Bryan Anderson)
* ASTERISK-23266 - [patch]pjsip_cli: Memory leak in
ast_sip_cli_print_sorcery_objectset (Reported by George Joseph)
* ASTERISK-19499 - ConfBridge MOH is not working for transferee
after attended transfer (Reported by Timo Teräs)
* ASTERISK-23261 - [patch]Output mixup in
${CHANNEL(rtpqos,audio,all)} (Reported by rsw686)
* ASTERISK-23279 - [patch]Asterisk doesn't support the dynamic
payload change in rtp mapping in the 200 OK response (Reported
by NITESH BANSAL)
* ASTERISK-23141 - Asterisk crashes on Dial(), in
pbx_find_extension at pbx.c (Reported by Maxim)
* ASTERISK-23336 - Asterisk warning Don't know how to indicate
condition 33 on ooh323c on outgoing calls from H323 to SIP peer
(Reported by Alexander Semych)
* ASTERISK-23320 - Preloading pbx_config.so with a CustomPresence
hint defined results in crash (Reported by xrobau)
* ASTERISK-23265 - Preloading Certain Modules in Asterisk 12
causes a core dump (Reported by Andrew Nagy)
* ASTERISK-23287 - res_pjsip_refer: Crash during attended transfer
when attended-transferer_second channel is NULL (Reported by
Matt Jordan)
* ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
to minrate=2400, then res_fax refuse to load (Reported by David
Brillert)
* ASTERISK-23135 - Crash - segfault in ast_channel_hangupcause_set
- probably introduced in 11.7.0 (Reported by OK)
* ASTERISK-23323 - [patch]chan_sip: missing p-owner checks in
handle_response_invite (Reported by Walter Doekes)
* ASTERISK-23406 - [patch]Fix typo in sip show peer (Reported by
ibercom)
* ASTERISK-22911 - [patch]Asterisk fails to resume WebRTC call
from hold (Reported by Vytis ValentinaviÄius)
* ASTERISK-23104 - Specifying the SetVar AMI without a Channel
cause Asterisk to crash (Reported by Joel Vandal)
* ASTERISK-21930 - [patch]WebRTC over WSS is not working.
(Reported by John)
* ASTERISK-23383 - Wrong sense test on stat return code causes
unchanged config check to break with include files. (Reported by
David Woolley)
* ASTERISK-20149 - Crash when faxing SIP to SIP with strictrtp set
to yes (Reported by Alexandr Gordeev)
* ASTERISK-23258 - Target_uri for LiveRecording model in ARI
(Reported by Ben Merrills)
* ASTERISK-17523 - Qualify for static realtime peers does not work
(Reported by Maciej Krajewski)
* ASTERISK-23204 - Device state cache requires improvement
(Reported by Mark Michelson)
* ASTERISK-23092 - cli: pjsip show endpoint endpoint shows
allow/disallow codecs the same (Reported by Dan Jenkins)
* ASTERISK-21406 - [patch] chan_sip deadlock on monlock between
unload_module and do_monitor (Reported by Corey Farrell)
* ASTERISK-23210 - Security: Remote crash in res_pjsip. (Reported
by Joshua Colp)
* ASTERISK-23373 - [patch]Security: Open FD exhaustion with
chan_sip Session-Timers (Reported by Corey Farrell)
* ASTERISK-23340 - Security Vulnerability: stack allocation of
cookie headers in loop allows for unauthenticated remote denial
of service attack (Reported by Matt Jordan)
* ASTERISK-23020 - PJSip - Multihomed machine returning wrong IP
address (Reported by xrobau)
* ASTERISK-23311 - Manager - MoH Stop Event fails to show up when
leaving Conference (Reported by Benjamin Keith Ford)
* ASTERISK-23295 - ARI: ChannelEnteredBridge event not delivered
to client during bridge move operation (Reported by Matt Jordan)
* ASTERISK-23444 - Playback and Record events not
[asterisk-users] Help with a bug
Dear friends I filed a bug https://issues.asterisk.org/jira/browse/ASTERISK-23656 but I am wondering if somebody can figure a workaround. I am stuck trying to deliver an application. The case is this: A Record is executed and an immediate Playback follows. Asterisk returns an error, saying that the file does not exist, but a few seconds later, it does. It does not help if after the Record application I do SHELL(sync). Asterisk has not flushed the file out to the OS and it already returned. Maybe the application record should have a parameter about this behavior. For some application is fine, for some others is not. -- _ -- 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] Open Source Asterisk Polling Solution
On Tue, 22 Apr 2014, A J Stiles wrote: ...so absolutely *do not* pay money for a solution, and *do* insist on the Source Code and Modification Rights. Even an obvious and simple solution has value if it exceeds the OP's skill set or the value of his time to implement. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Help with a bug
How many seconds later does the file show up? Can you just throw in a Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even a second or two of delay be an issue (or does it still not work)? -Josh On Wed, Apr 23, 2014 at 2:23 PM, CDR vene...@gmail.com wrote: Dear friends I filed a bug https://issues.asterisk.org/jira/browse/ASTERISK-23656 but I am wondering if somebody can figure a workaround. I am stuck trying to deliver an application. The case is this: A Record is executed and an immediate Playback follows. Asterisk returns an error, saying that the file does not exist, but a few seconds later, it does. It does not help if after the Record application I do SHELL(sync). Asterisk has not flushed the file out to the OS and it already returned. Maybe the application record should have a parameter about this behavior. For some application is fine, for some others is not. -- _ -- 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] Help with a bug
As a second possible solution, instead of Record, could you use MixMonitor, then run StopMixMonitor and THEN do your Playback? That should definitely make sure the recording file is closed and the file handle released. -Josh On Wed, Apr 23, 2014 at 2:29 PM, Josh Metzger joshdmetz...@gmail.comwrote: How many seconds later does the file show up? Can you just throw in a Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even a second or two of delay be an issue (or does it still not work)? -Josh On Wed, Apr 23, 2014 at 2:23 PM, CDR vene...@gmail.com wrote: Dear friends I filed a bug https://issues.asterisk.org/jira/browse/ASTERISK-23656 but I am wondering if somebody can figure a workaround. I am stuck trying to deliver an application. The case is this: A Record is executed and an immediate Playback follows. Asterisk returns an error, saying that the file does not exist, but a few seconds later, it does. It does not help if after the Record application I do SHELL(sync). Asterisk has not flushed the file out to the OS and it already returned. Maybe the application record should have a parameter about this behavior. For some application is fine, for some others is not. -- _ -- 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] Open Source Asterisk Polling Solution
On Wed, 23 Apr 2014, Steve Edwards wrote: I tried database access in the dialplan using the mysql() application years ago, just to confirm I was right and I was :) What an ugly, messy, fragile dialplan. On Wed, 23 Apr 2014, Doug Lytle wrote: With FuncODBC this is no longer an issue. All of the query logic is handled outside of the dial plan. I took a look and it looks like a step in the right direction, kind of a 'prepared statement' approach and it gets all the ugly quoting nonsense out of the dialplan. The query statement may be out of the dialplan, but the logic of what to do with the returned values remains. The OP stated that he was going to 'will wire it up to the DNC' (the National Do Not Call Registry?) which sounds like a simple 'query the database to see if the key exists' kind of thing for which ODBC seems reasonable. This application should be expanded to include multiple databases so his callers can press 1 to be queued for an agent or 2 to be added to his client's private DNC database. While checking 2 databases is no big deal, a simple 'check-dnc' AGI can hide those details and yield a cleaner dialplan. As the application matures, there may be additional enhancements that would lean towards wishing he had started down the AGI road. If the target list includes (but is not limited to) members of a group (like a church) you could have a situation where the callee is on the DNC, but has opted-in so you have another database to consider. How about checking the database to see the last time they had 'waste they need picked up?' If the 'waste' is charitable donations of clothing or furniture, I suspect most people would be good with just a call or 2 per year. How about letting the 'donor' schedule the number of months until the next call? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Help with a bug
Doesn't MixMonitor use sox to combine the incoming and outgoing recordings? If so, I'd expect MixMonitor to add MORE delay, not less. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Metzger Sent: Wednesday, April 23, 2014 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with a bug As a second possible solution, instead of Record, could you use MixMonitor, then run StopMixMonitor and THEN do your Playback? That should definitely make sure the recording file is closed and the file handle released. -Josh On Wed, Apr 23, 2014 at 2:29 PM, Josh Metzger joshdmetz...@gmail.com wrote: How many seconds later does the file show up? Can you just throw in a Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even a second or two of delay be an issue (or does it still not work)? -Josh On Wed, Apr 23, 2014 at 2:23 PM, CDR vene...@gmail.com wrote: Dear friends I filed a bug https://issues.asterisk.org/jira/browse/ASTERISK-23656 but I am wondering if somebody can figure a workaround. I am stuck trying to deliver an application. The case is this: A Record is executed and an immediate Playback follows. Asterisk returns an error, saying that the file does not exist, but a few seconds later, it does. It does not help if after the Record application I do SHELL(sync). Asterisk has not flushed the file out to the OS and it already returned. Maybe the application record should have a parameter about this behavior. For some application is fine, for some others is not. -- _ -- 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] Open Source Asterisk Polling Solution
I think it's all a matter of personal taste. I think the logic for add to DNC is extremely trivial and would be more complicated with an AGI. You have your prompt playback/read, if they hit 1, head to the queue, if they hit 2, it's a single dialplan line to put the info into the database and then one more like to Playback a nice Goodbye message. You also add one additional line before calling out in that case - one query for the national DNC and one for the internal DNC, or you can get really fancy and setup your table with an extra column that denotes national or internal. While you're add it, throw in that extra column for next call scheduling... All that being said, you would probably be better off pre-processing before sending the phone numbers to the PBX so you've already scrubbed the call list to only people not on a DNC list and within the scheduled call-back time. That way you save the checks for each call outbound and only have the code for adding people to the DNC and scheduling the next call, though it still could be done either way (though in the case of not pre-processing, you're getting closer to making Asterisk into a dialer system and then you can get into fancy legal issues about using an autodialer when you accidentally call someone who doesn't want to be called and they complain (dependant on jurisdiction). -Josh On Wed, Apr 23, 2014 at 2:36 PM, Steve Edwards asterisk@sedwards.comwrote: On Wed, 23 Apr 2014, Steve Edwards wrote: I tried database access in the dialplan using the mysql() application years ago, just to confirm I was right and I was :) What an ugly, messy, fragile dialplan. On Wed, 23 Apr 2014, Doug Lytle wrote: With FuncODBC this is no longer an issue. All of the query logic is handled outside of the dial plan. I took a look and it looks like a step in the right direction, kind of a 'prepared statement' approach and it gets all the ugly quoting nonsense out of the dialplan. The query statement may be out of the dialplan, but the logic of what to do with the returned values remains. The OP stated that he was going to 'will wire it up to the DNC' (the National Do Not Call Registry?) which sounds like a simple 'query the database to see if the key exists' kind of thing for which ODBC seems reasonable. This application should be expanded to include multiple databases so his callers can press 1 to be queued for an agent or 2 to be added to his client's private DNC database. While checking 2 databases is no big deal, a simple 'check-dnc' AGI can hide those details and yield a cleaner dialplan. As the application matures, there may be additional enhancements that would lean towards wishing he had started down the AGI road. If the target list includes (but is not limited to) members of a group (like a church) you could have a situation where the callee is on the DNC, but has opted-in so you have another database to consider. How about checking the database to see the last time they had 'waste they need picked up?' If the 'waste' is charitable donations of clothing or furniture, I suspect most people would be good with just a call or 2 per year. How about letting the 'donor' schedule the number of months until the next call? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] Help with a bug
That's the case with Monitor (apparently), but MixMonitor grabs both ends of the call. On a system I ran with lots of MixMonitor recording, Asterisk renamed / moved the recording file when a call completed, and that happened without any delay at all. Only one file was created for the entire call. On Wed, Apr 23, 2014 at 2:39 PM, Eric Wieling ewiel...@nyigc.com wrote: Doesn't MixMonitor use sox to combine the incoming and outgoing recordings? If so, I'd expect MixMonitor to add MORE delay, not less. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] On Behalf Of Josh Metzger Sent: Wednesday, April 23, 2014 2:35 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help with a bug As a second possible solution, instead of Record, could you use MixMonitor, then run StopMixMonitor and THEN do your Playback? That should definitely make sure the recording file is closed and the file handle released. -Josh On Wed, Apr 23, 2014 at 2:29 PM, Josh Metzger joshdmetz...@gmail.com wrote: How many seconds later does the file show up? Can you just throw in a Wait() (maybe 1 or 2 seconds) and then do the Playback, or would even a second or two of delay be an issue (or does it still not work)? -Josh On Wed, Apr 23, 2014 at 2:23 PM, CDR vene...@gmail.com wrote: Dear friends I filed a bug https://issues.asterisk.org/jira/browse/ASTERISK-23656 but I am wondering if somebody can figure a workaround. I am stuck trying to deliver an application. The case is this: A Record is executed and an immediate Playback follows. Asterisk returns an error, saying that the file does not exist, but a few seconds later, it does. It does not help if after the Record application I do SHELL(sync). Asterisk has not flushed the file out to the OS and it already returned. Maybe the application record should have a parameter about this behavior. For some application is fine, for some others is not. -- _ -- 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 -- _ -- 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] Help with a bug
On Wed, 23 Apr 2014, CDR wrote: The case is this: A Record is executed and an immediate Playback follows. Asterisk returns an error, saying that the file does not exist, but a few seconds later, it does. A simple test: exten = *,n,record(foo.wav) exten = *,n,playback(foo) works as expected for me with Asterisk 11.8.1. I notice in the console log you uploaded, you have a file name of '180-industry:sln' The syntax for record says 'filename.format' not 'filename:format' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] 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
