[asterisk-users] Moving from res_sip to pjsip and simple bridge
Hi all, on my old Asterisk 14.x box i use queue for some offices. For example, in this scenario phone 5710 is ringing (after passing through a queue...) and 5349 answer using REFER: -- SIP/5349-0072 answered Local/SIP-5710@MemberConnector-0031;2 -- Local/SIP-5710@MemberConnector-0031;1 connected line has changed. Saving it until answer for SIP/5002-006e -- Local/SIP-5710@MemberConnector-0031;1 answered SIP/5002-006e -- Channel SIP/5349-0072 joined 'simple_bridge' basic-bridge -- Channel Local/SIP-5710@MemberConnector-0031;2 joined 'simple_bridge' basic-bridge -- Stopped music on hold on SIP/5002-006e -- Channel Local/SIP-5710@MemberConnector-0031;1 joined 'simple_bridge' basic-bridge <55d1ecc5-b251-4e8b-b0f3-ea08254c0ffd> -- Channel SIP/5002-006e joined 'simple_bridge' basic-bridge <55d1ecc5-b251-4e8b-b0f3-ea08254c0ffd> > 0xa081718 -- Probation passed - setting RTP source address to 172.20.xx.xx:60640 on new Asterisk 15.2 i decide to move to PJSIP but this functionality don't work and, on REFER, call dropped. Maybe there's something needs to be enabled or checked ? Michele -- Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena tel: 0577.(23)5000 - central...@unisi.it Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PJSIP issue - Syntax error exception when parsing
Hi all, i'm getting this error: [Feb 21 09:29:09] ERROR[1250]: pjproject:0 : sip_transport.c Error processing 396 bytes packet from UDP 193.x:5060 : PJSIP syntax error exception when parsing '' header on line 2 col 1: SIP/2.0 480 User 7000 not registered Via: SIP/2.0/UDP 193.x:5060;received=193.xx;rport=5060;branch=z9hG4bKPjb092b027-a5b9-4683-8652-c7fefc06ae29 From: ;tag=3d0a19e7-eabe-4446-84dd-43f02d831033 To: ;tag=24eb447e8d0b8b1e81ba6efb9d8649a2.ea35 Call-ID: defee3c7-e5ba-41ff-9be7-3c37e62437f2 CSeq: 22011 INVITE Content-Length: 0 -- end of packet. Asterisk 15.2.0 and PJSip 2.7.1 Tnx, Michele -- Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena tel: 0577.(23)5000 - central...@unisi.it Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callback on busy
Hi all,
i'm using Asterisk as a media box for a VoIP network based on OpenSIPS.
When an user phone is busy, call was forwarded to an asterisk ext:
; ===
; Voicemail on NOT AVAILABLE
; ===
exten => _VMR_.,1,Noop("from-voip: ${CALLERID(num)} ${EXTEN}")
exten => _VMR_.,n,Set(DID=${EXTEN:4})
exten => _VMR_.,n,Answer()
exten => _VMR_.,n,Wait(1)
exten => _VMR_.,n,GotoIf(${VM_INFO(${DID},exists)}?avail:unavail)
exten => _VMR_.,n(avail),Voicemail(${DID},u)
exten => _VMR_.,n,Hangup()
exten => _VMR_.,n(unavail),Playback(vm-theperson)
exten => _VMR_.,n,SayDigits(${DID});
exten => _VMR_.,n,Playback(vm-isunavail)
exten => _VMR_.,n,Read(digit,vm-tocallback,1,,1,5)
exten => _VMR_.,n,Gotoif($["${digit}" = "2"]?:skip,1,5)
exten => _VMR_.,n,Noop("Add callback for ${DID} from ${CALLERID(num)}")
exten => _VMR_.,n,AGI(callback,${DID},${CALLERID(num)})
exten => _VMR_.,n,Playback(goodbye)
exten => _VMR_.(skip),n,Hangup()
when a vocal message asks to press "2" to add a callback when called
users return free, using an AGI script that create a .call file:
#!/usr/bin/php -q
\n");
fputs($cf,"MaxRetries: 100\n");
fputs($cf,"RetryTime: 30\n");
fputs($cf,"Archive: Yes\n");
fputs($cf,"SetVar: CALLER=$caller\n");
fputs($cf,"SetVar: CALLED=$called\n");
fclose($cf);
?>
periodically, Asterisk try to call CALLED user using CB_ routine:
; ===
; Callback
; ===
exten => _CB_.,1,Set(FROM=${CALLER})
exten => _CB_.,n,Set(TO=${CALLED})
exten => _CB_.,n,Noop("callback: from ${FROM} to ${TO}")
exten => _CB_.,n,Set(CALLERID(all)="CB ${FROM} <${FROM}>")
exten => _CB_.,n,Dial(SIP/voip-trunk/${TO},10) ; Check if called is
available and, if answer, transfer to caller (TODO)...
exten => _CB_.,n,Hangup()
But, as you can see, Callback routine wasn't done because i'm unable to
figure out how i can do that. I need that Asterisk call CALLED user and,
when answered, start calling CALLER.
It's possibile ?
Thanks, Michele
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[asterisk-users] _FAX_. extension refuses to work !
Hi all,
my dialplan is:
/;
==//
//; FROM VOIP//
//;
==//
//
//[from-voip]//
//include => default//
//
//[default]
/
/; FAXes//
//exten => _FAX_.,1,Noop("from-voip: FAX ${CALLERID(num)} ${EXTEN}")//
//exten => _FAX_.,n,Set(DID=${EXTEN:4})//
//exten => _FAX_.,n,Goto(fax-services,s,1)/
but on a call directed to, es. FAX_3700 i got:
[Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309
handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension
'FAX_3700' rejected because extension not found in context 'from-voip'.
[Nov 30 11:38:30] NOTICE[5462][C-0027]: chan_sip.c:26309
handle_request_invite: Call from 'voip-trunk' (xxx:5060) to extension
'FAX_3700' rejected because extension not found in context 'from-voip'.
Other extension like _IVR_ or _VMR_ works perfeclty and are defined in
the same manner.
Maybe _FAX was a reserved keyword ?
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it
Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di
Ateneo, http://www.faq.unisi.it
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Re: [asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data
Yes, it works ! Thanks :-) Michele On 30/11/2016 10:19, Jonathan H wrote: > I think it might be related to this? > https://issues.asterisk.org/jira/browse/ASTERISK-26391 > > I think I remember having to edit logger.conf - this is what mine > looks like now: > console => notice,warning,error > messages => notice,warning,error > > Try that, restart asterisk and see if it works :) -- Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena tel: 0577.(23)5000 - central...@unisi.it Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 14.2 CLI don't show debug/verbose data
Hi all, after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) don't show what's happens. I've trying setting debug and verbose to 100 but nothing, no show. All commands works as expected but i can't what's happens on my asterisk server. asterisk*CLI> core show settings PBX Core settings - Version: 14.2.0 Build Options: LOADABLE_MODULES, BUILD_NATIVE, OPTIONAL_API Maximum calls: 30 (Current 0) Maximum open file handles: 1024 Root console verbosity: 100 Current console verbosity: 100 Debug level: 100 Maximum load average:0.90 Minimum free memory: 1 MB Startup time:09:07:33 Last reload time:09:07:33 System: Linux/3.16.0-4-686-pae built by root on i686 2016-11-28 14:50:24 UTC System name: Default language:en Language prefix: Enabled User name and group: / Executable includes: Disabled Transcode via SLIN: Enabled Transmit silence during rec: Disabled Generic PLC: Enabled Min DTMF duration:: 80 RTP dynamic payload types: 96-127 * Subsystems - Manager (AMI): Disabled Web Manager (AMI/HTTP): Disabled Call data records: Disabled Realtime Architecture (ARA): Disabled * Directories - Configuration file: /etc/asterisk/asterisk.conf Configuration directory: /etc/asterisk Module directory:/usr/lib/asterisk/modules Spool directory: /var/spool/asterisk Log directory: /var/log/asterisk Run/Sockets directory: /var/run/asterisk PID file:/var/run/asterisk/asterisk.pid VarLib directory:/var/lib/asterisk Data directory: /var/lib/asterisk ASTDB: /var/lib/asterisk/astdb IAX2 Keys directory: /var/lib/asterisk/keys AGI Scripts directory: /var/lib/asterisk/agi-bin Any hint ? Michele -- Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena tel: 0577.(23)5000 - central...@unisi.it Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 13 Realtime Voicemail frustrating issue
Hi all, i'm experiencing a really frustrating issue with my Asterisk 13.7.2 with realtime configuration on MySQL and Voicemail. Here's res_config_mysql.conf: /[default]// //dbhost = 192.168.1.1// //dbname = asterisk// //dbuser = asterisk// //dbpass = [x]// //dbport = 3306// //requirements=warn ; or createclose or createchar// / extconfig.conf: /[settings]// //sipusers => mysql,default,sipusers// //sippeers => mysql,default,sipusers// //sipregs => mysql,default,sipregs// //voicemail => mysql,default,vmusers// //meetme => mysql,default,meetme// / on Asterisk console: /asterisk*CLI> realtime mysql status // //default connected to asterisk@192.168.1.1, port 3306 with username asterisk for 56 minutes.// //asterisk*CLI> / "vmusers" table on MySQL: uniqueid <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60uniqueid%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> customer_id <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60customer_id%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> context <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60context%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> mailbox <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60mailbox%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> password <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60password%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> fullname <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60fullname%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> email <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60email%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> pager <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60pager%60+ASC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> stamp <http://voip.unisi.it/phpmyadmin/sql.php?db=asterisk&table=vmusers&sql_query=SELECT+%2A+FROM+%60vmusers%60%0AORDER+BY+%60vmusers%60.%60stamp%60+DESC&session_max_rows=25&token=81771f45cae5714ad1fac75365e0e494> 50025002default 5002AAA /NULL/ -00-00 00:00:00 50055005default 5005bbb /NULL/ -00-00 00:00:00 50185018default 5018ccc /NULL/ -00-00 00:00:00 50075007default 5007s /NULL/ -00-00 00:00:00 *BUT* when i type, on Asterisk console: / //asterisk*CLI> voicemail show zones // //There are no voicemail zones currently defined// //Command 'voicemail show zones ' failed.// //asterisk*CLI> / the same, of course, for "show users default". And whet i try to access a mailbox, i get a "Invalid password". Any hints ? Please, i'm really frustrated ! Michele -- Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena tel: 0577.(23)5000 - central...@unisi.it Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it -- _ -- 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] Call hangup on transfer when originated from a Queue
27362]: 4796fc0e1fca050c0367076b49ec1...@voip.unisi.it - Route RELAY REFER To: 2169, From: 5002, RURI: sip:2169@172.20.1.5:5060 /usr/sbin/opensips[27359]: 4796fc0e1fca050c0367076b49ec1...@voip.unisi.it - Route RELAY NOTIFY To: 5002, From: 2169, RURI: sip:5002@172.20.1.47:37496 /usr/sbin/opensips[27361]: d59a893ecb225520 - Route RELAY BYE To: 2169, From: 5000, RURI: sip:2169@172.20.1.4:5060 /usr/sbin/opensips[27362]: 4796fc0e1fca050c0367076b49ec1...@voip.unisi.it - Route RELAY NOTIFY To: 5002, From: 2169, RURI: sip:5002@172.20.1.47:37496 /usr/sbin/opensips[27361]: 313435303331343237333731-iqp91rwhq7h6 - Route RELAY NOTIFY To: 5009, From: 5002, RURI: sip:5009@172.20.1.215:32768 /usr/sbin/opensips[27361]: 4796fc0e1fca050c0367076b49ec1...@voip.unisi.it - Route RELAY BYE To: 2169, From: 5002, RURI: sip:2169@172.20.1.5:5060 Thanks for any help or suggestion ! Michele -- Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena tel: 0577.(23)5000 - central...@unisi.it Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it -- _ -- 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] Call diversion
Hi all,
i'm trying to setup a function like "secretary/director": when an user
call director number (eg. 5000), the call were firstly diverted to
secretary (5001). At this point, when secretary answer, can decide to
transfer back the call to director (5000).
Because i'm using OpenSIPS as SIP router, when this function is needed i
did a special "extension" like DIVERT_[from]_[to]:
; On DIVERT
exten => _DIVERT_.,1,Noop("from-voip: DIVERT ${CALLERID(num)} -
${EXTEN}") ; Example: DIVERT_5000_5001
exten => _DIVERT_.,n,Set(DIVFrom=${CUT(EXTEN,_,2)})
exten => _DIVERT_.,n,Set(DIVTo=${CUT(EXTEN,_,3)})
exten => _DIVERT_.,n,Noop("Divert ${CALLERID(num)} from ${DIVFrom} to
${DIVTo}")
exten => _DIVERT_.,n,Macro(services-divert) // Divert
exten => _DIVERT_.,n,Hangup()
and a macro like:
[macro-services-divert] ; Servizio direttore-segretaria
exten => s,1,Noop("Divert from ${CALLERID(num)}")
same => n,Answer
same => n,Playback(msg/msg_attendereufficiodesiderato)
same => n,Dial(SIP/voip-trunk/${DIVTo},30)
It works *BUT* when from the secretary phone the call were forwarded
back to director (and, at this point, director phone must ring !), the
call stall with an error on OpenSIPS like:
In-Dialog NOTIFY from [my asterisk box ip]
(callid=707902a659eb005b141a9491195bd3cf@voip) is not valid according to
dialog
Any hits ?
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it
Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di
Ateneo, http://www.faq.unisi.it
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Re: [asterisk-users] Realtime Voicemail MWI
Hi Nick, did you set-up also Voicemail boxes in Realtime ? Michele Il 16/09/2015 22:44, Nick Olsen ha scritto: > Greetings All, Regarding this archived > post. > http://lists.digium.com/pipermail/asterisk-users/2014-November/285169.html > > Did anyone ever find an solution to this? I've got a new box running > 13.3.0 with the exact same issue. > > For those that don't read the link. > > I've got SIP Peers in realtime. All with a mailbox set. 98% of the > time, These are loaded into asterisk without the mailbox info. Leading > to "Received SIP subscribe for peer without mailbox" notices. And > non-working MWI. > > Occasionally, It just works. But only on a peer or two at a time. And > it'll stop working after a few minutes. > > Any ideas? Thanks > > -- Michele Pinassi Responsabile Telefonia di Ateneo Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena tel: 0577.(23)5000 - fax: 0577.(23)2053 Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di Ateneo, http://www.faq.unisi.it signature.asc Description: OpenPGP digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenSIPS, Asterisk and LocalAgents for Queues
Hi all,
i'm build and using a voip pbx system using OpenSIPS as a router (i need
to serve thousand of users...) and an Asterisk server as media box, for
IVR, queues and so on.
I've a PATTON PSTN GW (172.20.1.4), the VoIP OpenSIPS ROUTER
(172.20.1.2) andn
In queues, because i've some troubles telling Asterisk which users are
online and available, i decide to use LocalAgent way to force calls to
every agents. For example, in queue.conf i have:
[operator-phone-queue]
music = queue-default
strategy = linear
context = ivr-services ; Here we go when the caller presses a single
digit, while in the queue
timeout = 15
wrapuptime = 10
announce-frequency = 30
announce-holdtime = yes
joinempty = yes
member => Local/SIP-5002@MemberConnector,1
member => Local/SIP-5023@MemberConnector,2
and in extensions.conf:
[MemberConnector]
exten => _[A-Za-z0-9].,1,Verbose(2,Connecting ${CALLERID(all)} to Agent
at ${EXTEN})
same => n,Set(QueueMember=${FILTER(A-Za-z0-9\-,${EXTEN})})
same => n,Set(Technology=${CUT(QueueMember,-,1)})
same => n,Set(Device=${CUT(QueueMember,-,2)})
same => n,Noop("MemberConnector: calling queue member
${Technology}/voip-trunk/${Device}")
same => n,Dial(${Technology}/voip-trunk/${Device},30)
same => n,Hangup()
That way works well *BUT* i have a problem with RTP audio flow, because
when, for example, i call from 4999 to the queue and 5002 or 5023
answers the call, i got no audio from 5002 to 4999 (but i hear sounds
from 4999 to 5002). The SIP signalling was this:
INVITE sip:5002@172.20.1.47:57907 SIP/2.0.
Via: SIP/2.0/UDP 172.20.1.2:5060;branch=z9hG4bKa165.92c040a1.0.
Via: SIP/2.0/UDP
172.20.1.5:5060;rport=5060;received=172.20.1.5;branch=z9hG4bK47310f8d.
Max-Forwards: 69.
From: ;tag=as1e28f247.
To: .
Call-ID: 252126f32e04b0364360b6d65c7dba1f@.
CSeq: 104 INVITE.
User-Agent: Asterisk PBX 11.13.1~dfsg-2+b1.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGE.
Supported: replaces, timer.
Content-Type: application/sdp.
Content-Length:240.
.
v=0.
o=root 862552143 862552145 IN IP4 172.20.1.5.
s=Asterisk PBX 11.13.1~dfsg-2+b1.
c=IN IP4 172.20.1.5.
t=0 0.
m=audio 16660 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
[...]
SIP/2.0 200 Ok.
Via: SIP/2.0/UDP
172.20.1.5:5060;rport=5060;received=172.20.1.5;branch=z9hG4bK47310f8d.
From: ;tag=as1e28f247.
To: "Michele" ;tag=l3f2mwdv8j.
Call-ID: 252126f32e04b0364360b6d65c7dba1f@.
CSeq: 104 INVITE.
User-Agent: snom760/8.7.5.17.
Contact: ;reg-id=1.
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
PRACK, MESSAGE, INFO, UPDATE.
Allow-Events: talk, hold, refer, call-info.
Supported: timer, 100rel, replaces, from-change.
Content-Type: application/sdp.
Content-Length: 218.
.
v=0.
o=root 1421125882 1421125885 IN IP4 172.20.1.47.
s=call.
c=IN IP4 172.20.1.47.
t=0 0.
m=audio 60670 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=ptime:20.
a=sendrecv.
I think that the problem was the 172.20.1.5 (Asterisk box) as RTP
endpoint and not 172.20.1.4 (Patton GW, where call 4999 was originated).
Just to be more clear, the flow is:
[PSTN Net 4999]>[PATTON GW | 172.20.1.4]>[OpenSIPS
172.20.1.2]--->[Asterisk BOX (Queues) | 172.20.1.5]>[OpenSIPS
172.20.1.2]>(ring 5002)>(answer 5002)--->(Call established but
no audio)
So, there's a solution ? Hints ?
Thanks, Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi
di Siena
tel: 0577.(23)5000 - fax: 0577.(23)2053
Per trovare una soluzione rapida ai tuoi problemi tecnici
consulta le FAQ di Ateneo, http://www.faq.unisi.it
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