[asterisk-users] asterisk-java is dead?
Hello Everyone, I am trying to make use of asterisk-java live and had some questions for the mailing list however, it does not seem like it's an active mailing list? Is the project dead? Thanks, Nick -- _ -- 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] OpenSIPS, Asterisk and LocalAgents for Queues
Hi, I hope you already have fixed it . In case you didnt then here are my thoughts. Looking at the flow and keeping in mind that all devices are in same subnet you should never get one way audio issue since OpenSIP is not playing with SDP so Asterisk and PGW should just be able to have two way audio all the time and so you should debug the B leg going out to the endpoint. BR, Sammy. -- _ -- 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] Found audio description format L16 for ID 98 No compatible codecs, not accepting this ?
i am trying to receive a call from freeswitch without transcoding , asterisk and freeswitch are installed on same machine in asterisk cli with sip set debug on v=0 o=FreeSWITCH 1442495774 1442495775 IN IP4 127.0.0.1 s=FreeSWITCH c=IN IP4 127.0.0.1 t=0 0 m=audio 28840 RTP/AVP 98 13 a=rtpmap:98 L16/16000 a=ptime:20 Found RTP audio format 98 Found RTP audio format 13 Found audio description format L16 for ID 98 chan_sip.c:10556 process_sdp: No compatible codecs, not accepting this offer! is it possible to receive this call and pass it to chan_dongle ?? -- _ -- 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] Calls to Ring Group not working. FreePBX.
Hi All, Hi All, I am trying to create an Inbound route destined to a Ring Group through a SIP trunk. I am able to call the extensions directly, but unable to call a Ring Group or an IVR through the Inbound Route config. I am really not sure, what i am missing. When the DID for the IVR or Ring Group is called, getting the message from the Asterisk that "the call cannot be completed, please check your number". I am doing the configuration using FreePBX and the Asterisk version is 12. The Inbound Route configuration for the IVR :- 1. DID Number : 2000 2. Ring Groups : RG<600> SIP Peer details :- host=20.1.1.170 type=friend port=5060 nat=no disallow=all allow=ulaw,alaw qualify=yes canreinvite=yes context=from-trunk When 2000, is dialled, the DID in the SIP Invite is the same, but still getting the error message. SIP Logs :- Invite to the DID 2000 for Ring Group > 100 Trying <- 183 Session Progess <- (Playing the error message) -- <--- SIP read from UDP:20.1.1.170:5060 ---> INVITE sip:2000@20.1.1.58:5060 SIP/2.0 Via: SIP/2.0/UDP 20.1.1.170:5060;branch=z9hG4bK3b3995664c39c From: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395 To: Date: Fri, 11 Sep 2015 14:06:41 GMT Call-ID: 52087400-5f21dff1-354b2-aa010114@20.1.1.170 Supported: timer,resource-priority,replaces Min-SE: 1800 User-Agent: Cisco-CUCM10.5 Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY CSeq: 101 INVITE Expires: 180 Allow-Events: presence Supported: X-cisco-srtp-fallback,X-cisco-original-called Cisco-Guid: 1376285696-065536-002594-2852192532 Session-Expires: 1800 P-Asserted-Identity: Remote-Party-ID: ;party=calling;screen=yes;privacy=off Contact: ;bfcp Max-Forwards: 69 Content-Type: application/sdp Content-Length: 198 v=0 o=CiscoSystemsCCM-SIP 787014 1 IN IP4 20.1.1.170 s=SIP Call c=IN IP4 20.1.1.170 t=0 0 m=audio 25986 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 <-> --- (22 headers 9 lines) --- Sending to 20.1.1.170:5060 (no NAT) Sending to 20.1.1.170:5060 (no NAT) Using INVITE request as basis request - 52087400-5f21dff1-354b2-aa010114@20.1.1.170 Found peer '2723' for '2723' from 20.1.1.170:5060 Found RTP audio format 0 Found RTP audio format 101 Found audio description format PCMU for ID 0 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 20.1.1.170:25986 Looking for 2000 in from-internal (domain 20.1.1.58) list_route: hop: <--- Transmitting (NAT) to 20.1.1.170:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 20.1.1.170:5060 ;branch=z9hG4bK3b3995664c39c;received=20.1.1.170;rport=5060 From: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395 To: Call-ID: 52087400-5f21dff1-354b2-aa010114@20.1.1.170 CSeq: 101 INVITE Server: FPBX-12.0.76(11.19.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Length: 0 <> Audio is at 16598 Adding codec 13 (ulaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP <--- Transmitting (NAT) to 20.1.1.170:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 20.1.1.170:5060 ;branch=z9hG4bK3b3995664c39c;received=20.1.1.170;rport=5060 From: ;tag=787014~4ab333c0-314e-1172-16a8-eca8c1530263-31444395 To: ;tag=as3e6a1653 Call-ID: 52087400-5f21dff1-354b2-aa010114@20.1.1.170 CSeq: 101 INVITE Server: FPBX-12.0.76(11.19.0) Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: Content-Type: application/sdp Require: timer Content-Length: 228 v=0 o=root 881046367 881046367 IN IP4 20.1.1.58 s=Asterisk PBX 11.19.0 c=IN IP4 20.1.1.58 t=0 0 m=audio 16598 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv = Anything i am missing here ? Also please let me know, if you need any other logs to help me in this. Thanks a lot ! Agasthian P -- _ -- 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 AMI events filtering
Sam Based on my experience you need to write a middle tier that has what you want exposed to the users.. AMI was not really designed to offer direct multi-tenant access. That is for your middle tier to handle. Thanks Bryant Zimmerman (ZK Tech Inc.) 616-855-1030 Ext. 2003 From: "Sam Basan" Sent: Thursday, September 17, 2015 7:21 AM To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: [asterisk-users] Asterisk AMI events filtering Hi folks, I have one server with multiple companies (multi-tenant). >From AMI I get all events of all extensions so any one that connect can see other extensions, from different company (context). How can I limit specific user to get just specific context? Sam -- _ -- 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] Realtime Voicemail MWI
Yes, They are. Nick Olsen Network Operations (855) FLSPEED x106 From: "Michele Pinassi" Sent: Thursday, September 17, 2015 3:07 AM To: asterisk-users@lists.digium.com Subject: 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 -- _ -- 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] I want to store cdr into database
It is very simple, asterisk can log cdrs automatically by configuring cdr_mysql.conf. All you need to create a mysql table along with proper read/write permissions. You can find the cdr table schema from the below link. https://wiki.asterisk.org/wiki/display/AST/MySQL+CDR+Backend Regards, Muhammad Faheem On Thu, Sep 17, 2015 at 3:21 PM, Amelye Chatila wrote: > I have asterisk 13.5 configured with a simple dial plan, 3 SIP clients two > Laptops and smartphone with softphones installed. Now I am trying to store > cdr into a database but not able to make a connection of ODBC drivers to > MySQL is there an option or anything. Thanks in advance > > My configuration:: > *sip.conf* > > [general] > trasport=udp ;Data format | sample commennt > > [template01](!) > type=friend > context=from-internal > host=dynamic > disallow=all > allow=ulaw > context=from-internal > secret=unsecurepassword > > [6001](template01) > > [7001](template01) > bindport=6050 > > > *extensions.conf* > > [from-internal] > exten => 7001,1,Dial(SIP/7001,30) > exten => 6001,1,Dial(SIP/6001,30) > > -- > _ > -- 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] how to get an info from "To:" header?
Le 17/09/2015 12:37, Дорофеев Сергей a écrit :
Hello list!
Hello
Sorry for kinda dumb question, I guess, but I have too little time to
research it by myself.
I have a SIP packet, which looks like this:
<--- SIP read from UDP:10.186.0.38:5060 --->
INVITE sip:XXX@10.186.35.98:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.186.0.38:5060;branch=z9hG4bKh4utm43008vheqk093b0.1
Call-ID: ba9vp4zsbbsfi0vagdafg0vpzpp0z9wh@SoftX3000
From: ;tag=zwbzfehp-CC-22
To:
CSeq: 1 INVITE
Contact:
Min-SE: 90
Session-Expires: 300
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R011
Diversion:
;reason=unconditional;counter=1
Supported: 100rel,timer
Max-Forwards: 69
Content-Length: 338
Content-Type: application/sdp
Priority: urgent
I need to use info from fields “To:” and “Contact:” later in my
dialplan. I belive, I have to do something like “exten =>
_/X.,1,Set(VAR=${WHAT/_SHOULD_I_TYPE_HERE?})”
Sample:
exten => s,1,Set(__DIALEDNUMBER=${SIP_HEADER(TO):5})
exten => s,n,Set(__DIALEDNUMBER=${CUT(DIALEDNUMBER,@,1)})
...
Regards
--
Daniel
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk AMI events filtering
Hi folks, I have one server with multiple companies (multi-tenant). >From AMI I get all events of all extensions so any one that connect can see other extensions, from different company (context). How can I limit specific user to get just specific context? Sam -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to get an info from "To:" header?
Hello list!
Sorry for kinda dumb question, I guess, but I have too little time to research
it by myself.
I have a SIP packet, which looks like this:
<--- SIP read from UDP:10.186.0.38:5060 --->
INVITE sip:XXX@10.186.35.98:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.186.0.38:5060;branch=z9hG4bKh4utm43008vheqk093b0.1
Call-ID: ba9vp4zsbbsfi0vagdafg0vpzpp0z9wh@SoftX3000
From: ;tag=zwbzfehp-CC-22
To:
CSeq: 1 INVITE
Contact:
Min-SE: 90
Session-Expires: 300
Allow:
INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
User-Agent: Huawei SoftX3000 V300R011
Diversion: ;reason=unconditional;counter=1
Supported: 100rel,timer
Max-Forwards: 69
Content-Length: 338
Content-Type: application/sdp
Priority: urgent
I need to use info from fields "To:" and "Contact:" later in my dialplan. I
belive, I have to do something like "exten =>
_X.,1,Set(VAR=${WHAT_SHOULD_I_TYPE_HERE?})"
Could you kindly help me, please?
WBR,
Dorofeev Sergey
Это электронное сообщение и любые документы, приложенные к нему, содержат
конфиденциальную информацию и предназначены исключительно для использования
сотрудниками компании, физическим или юридическим лицом, которому они
адресованы. Уведомляем Вас о том, что если это сообщение не предназначено Вам,
использование, копирование, распространение информации, содержащейся в
настоящем сообщении, а также осуществление любых действий на основе этой
информации, не допускается. Если вы получили это электронное сообщение по
ошибке, пожалуйста, свяжитесь с отправителем и удалите электронное сообщение и
любые файлы, передаваемые с ним, с компьютера незамедлительно. Спасибо.
--
_
-- 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] I want to store cdr into database
I have asterisk 13.5 configured with a simple dial plan, 3 SIP clients two Laptops and smartphone with softphones installed. Now I am trying to store cdr into a database but not able to make a connection of ODBC drivers to MySQL is there an option or anything. Thanks in advance My configuration:: *sip.conf* [general] trasport=udp ;Data format | sample commennt [template01](!) type=friend context=from-internal host=dynamic disallow=all allow=ulaw context=from-internal secret=unsecurepassword [6001](template01) [7001](template01) bindport=6050 *extensions.conf* [from-internal] exten => 7001,1,Dial(SIP/7001,30) exten => 6001,1,Dial(SIP/6001,30) -- _ -- 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] Update peer IP address
Am 16. September 2015 18:48:16 MESZ, schrieb Daniel Heckl : >Sebastian, > >If I have understood you correctly, the SIP communication is now via >NAT instead forwarded ports. For safety, it is much better. > >I think it is not because of a UDP timeout, but rather because of a NAT >timeout. For this is "qualify" exactly the right thing to let the NAT >port opened. > >Daniel Hi Daniel, Not quite. Asterisk is running on an Openwrt router. So Asterisk is listening on a public IP. No NAT involved, no port forwarding. Openwrt tracks the UDP connection for 180s (default). "qualify" keeps the connection alive (every 120s). Without "qualify" inbound calls wouldn't work starting 180s after the registration, until after another 300s, when Asterisk registers again (provider requires a registration expiry >480s). Regards, Sebastian -- _ -- 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] 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
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
