[asterisk-users] Dynamic Music on Hold
Hello everyone, I am trying to activate Music On Hold using DB on Asterisk 13. It works fine but in order to use new Music On hold definitions I have to reload the moh module. - The following is my configuration in extconfig.conf - I added the following line: musiconhold.conf => mysql,asterisk,bit_ast_config - The following is the table in the database: mysql> select * from bit_ast_config; +++-++---+---+---+--+ | id | cat_metric | var_metric | commented | filename| category | var_name | var_val | +++-++---+---+---+--+ | 2 | 0 | 0 | 0 | musiconhold.conf | yaron | directory | moh | | 3 | 0 | 0 | 0 | musiconhold.conf | yaron | mode | files | | 10 | 0 | 0 | 0 | musiconhold.conf | yaron1| directory | moh | | 11 | 0 | 0 | 0 | musiconhold.conf | yaron1| mode | files | +++-++---+---+---+--+ Is there a way to do automatically add new moh definitions without reloading the moh module? Thanks, Yaron. -- _ -- 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 does not listed to port 5060
Hi Friends, I encountered a strange issue. I am running Asterisk 11.8.1 on Cent OS with no firewall running. It has 3 NIC interfaces. in my sip.conf I have allowguest=yes bindaddr=0.0.0.0 udpbindaddr = 0.0.0.0 But my Asterisk instance does not pick the call at all. When I check the listening apps using lsof -i I get asterisk 3046 asterisk7u IPv4 1191172 0t0 TCP *:5038 (LISTEN) asterisk 3046 asterisk 10u IPv4 1191186 0t0 UDP *:sip asterisk 3046 asterisk 11u IPv4 1191187 0t0 TCP *:sip (LISTEN) asterisk 3046 asterisk 13u IPv4 1191196 0t0 UDP *:iax asterisk 3046 asterisk 15u IPv4 1191199 0t0 UDP *:commplex-main asterisk 3046 asterisk 16u IPv4 1191201 0t0 UDP *:4520 asterisk 3046 asterisk 19u IPv4 1191232 0t0 TCP localhost:5038->localhost:43353 (ESTABLISHED) But I van see the SIP Invite that comes into server and I can ngrep it as U 10.85.0.24:5060 -> 10.25.172.10:5060 INVITE sip:+91712442211@10.25.172.10:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1. Content-Type: application/sdp. To: . Contact: . From: ;tag=5BD23246313536415F1CF602. P-Asserted-Identity: . Privacy: none. Supported: histinfo,100rel. Request-Disposition: no-fork. P-Early-Media: supported. History-Info: ;index=1, ;index=1.1. Max-Forwards: 69. Accept: application/sdp. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK. Call-ID: 1207028FDD81406CCD7E@GPAS_GWCS6_ipm_1_2_6. CSeq: 1 INVITE. Content-Length: 171. . v=0. o=- 1 1 IN IP4 10.85.0.24. s=-. t=0 0. m=audio 36740 RTP/AVP 8 101. c=IN IP4 10.85.0.24. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. U 10.85.0.24:5060 -> 10.25.172.10:5060 INVITE sip:+91712442211@10.25.172.10:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1. Content-Type: application/sdp. To: . Contact: . From: ;tag=5BD23246313536415F1CF602. P-Asserted-Identity: . Privacy: none. Supported: histinfo,100rel. Request-Disposition: no-fork. P-Early-Media: supported. History-Info: ;index=1, ;index=1.1. Max-Forwards: 69. Accept: application/sdp. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK. Call-ID: 1207028FDD81406CCD7E@GPAS_GWCS6_ipm_1_2_6. CSeq: 1 INVITE. Content-Length: 171. . v=0. o=- 1 1 IN IP4 10.85.0.24. s=-. t=0 0. m=audio 36740 RTP/AVP 8 101. c=IN IP4 10.85.0.24. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. U 10.85.0.24:5060 -> 10.25.172.10:5060 INVITE sip:+91712442211@10.25.172.10:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1. Content-Type: application/sdp. To: . Contact: . From: ;tag=5BD23246313536415F1CF602. P-Asserted-Identity: . Privacy: none. Supported: histinfo,100rel. Request-Disposition: no-fork. P-Early-Media: supported. History-Info: ;index=1, ;index=1.1. Max-Forwards: 69. Accept: application/sdp. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK. Call-ID: 1207028FDD81406CCD7E@GPAS_GWCS6_ipm_1_2_6. CSeq: 1 INVITE. Content-Length: 171. . v=0. o=- 1 1 IN IP4 10.85.0.24. s=-. t=0 0. m=audio 36740 RTP/AVP 8 101. c=IN IP4 10.85.0.24. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. U 10.85.0.24:5060 -> 10.25.172.10:5060 INVITE sip:+91712442211@10.25.172.10:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1. Content-Type: application/sdp. To: . Contact: . From: ;tag=5BD23246313536415F1CF602. P-Asserted-Identity: . Privacy: none. Supported: histinfo,100rel. Request-Disposition: no-fork. P-Early-Media: supported. History-Info: ;index=1, ;index=1.1. Max-Forwards: 69. Accept: application/sdp. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK. Call-ID: 1207028FDD81406CCD7E@GPAS_GWCS6_ipm_1_2_6. CSeq: 1 INVITE. Content-Length: 171. . v=0. o=- 1 1 IN IP4 10.85.0.24. s=-. t=0 0. m=audio 36740 RTP/AVP 8 101. c=IN IP4 10.85.0.24. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. U 10.85.0.24:5060 -> 10.25.172.10:5060 INVITE sip:+91712442211@10.25.172.10:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1. Content-Type: application/sdp. To: . Contact: . From: ;tag=5BD23246313536415F1CF602. P-Asserted-Identity: . Privacy: none. Supported: histinfo,100rel. Request-Disposition: no-fork. P-Early-Media: supported. History-Info: ;index=1, ;index=1.1. Max-Forwards: 69. Accept: application/sdp. Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,PRACK. Call-ID: 1207028FDD81406CCD7E@GPAS_GWCS6_ipm_1_2_6. CSeq: 1 INVITE. Content-Length: 171. . v=0. o=- 1 1 IN IP4 10.85.0.24. s=-. t=0 0. m=audio 36740 RTP/AVP 8 101. c=IN IP4 10.85.0.24. a=rtpmap:8 PCMA/8000. a=rtpmap:101 telephone-event/8000. a=ptime:20. U 10.85.0.24:5060 -> 10.25.172.10:5060 INVITE sip:+91712442211@10.25.172.10:5060;user=phone SIP/2.0. Via: SIP/2.0/UDP 10.85.0.24:5060;branch=z9hG4bK4l0vt3201860kmc5e6k0.1. Conte
Re: [asterisk-users] [OT] switches
On Fri, 20 Feb 2015 13:05:56 -0700, Harry McGregor wrote: > For a very basic setup it would work, but I would suggest POE at a > minimum, and vlan support if possible. thanks for the recomendations :) -Thufir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] switches
Hello, > Pardon, this might be off-topic. I'm reading: > > http://en.wikipedia.org/wiki/Network_switch > > For a setup of ~5 agents, would I be wrong in thinking that a generic 16 > port unmanaged switch would fit the bill? > > The first model to come up for me in an Amazon search is: > > http://support.netgear.com/product/fs116 > > > > Is this a reasonable choice? Would I be wrong in thinking that most any > Fast Ethernet switch would be fine for Asterisk? Yes, this kind of switches would work. VLAN and PoE support would obviously be better for convenience and security, but those are not mandatory. -- Bertrand LUPART -- _ -- 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] Queue PJSIP, not all contacts rings
Nick Awesome wrote:
Hay guys, have question.
When I do regular dial I use
$this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);
to get all contacts of current endpoint and so I dial to all phones
at once,
but if I exec QUEUE, I have just one phone rings, seems like it take
first one as Dial app by default, is there way to fix this?
There is no way to directly do this. The best option is to use a Local
channel into the dialplan which dials instead. Once answered everything
should fall into place.
--
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
--
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[asterisk-users] Queue PJSIP, not all contacts rings
Hay guys, have question.
When I do regular dial I use
$this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);
to get all contacts of current endpoint and so I dial to all phones at once,
but if I exec QUEUE, I have just one phone rings, seems like it take first one
as Dial app by default, is there way to fix this?
--
_
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Re: [asterisk-users] Queue PJSIP, not all contacts rings
Works, thank you!
> On Feb 23, 2015, at 7:11 PM, Joshua Colp wrote:
>
> Nick Awesome wrote:
>> Hay guys, have question.
>>
>> When I do regular dial I use
>> $this->AGI->get_fullvariable('${PJSIP_DIAL_CONTACTS('.$callObj.')}',false,true);
>>
>> to get all contacts of current endpoint and so I dial to all phones
>> at once,
>>
>> but if I exec QUEUE, I have just one phone rings, seems like it take
>> first one as Dial app by default, is there way to fix this?
>
> There is no way to directly do this. The best option is to use a Local
> channel into the dialplan which dials instead. Once answered everything
> should fall into place.
>
> --
> 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
--
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[asterisk-users] Question about Warning message
Starting with Asterisk 13.1 we are seeing this WARNING
messages a lot in our logs and console:
WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type
frames with SIP write)
We found that line in function "sip_write" inside "chan_sip.c".
In our previous version (11.2.1) we did not see those messages being printed
(same verbosity level). We compared both versions of the functions and see no
difference at all in the 'default' switch case that handles that. We
think/assume that that function is being called in
different places on each version (11.2-1 vs 13-1).
We also think it has to do with the asterisk receiving rtp packets with comfort
noise which is not supported by asterisk.
We would like to know what can we do about it to behave more like the version
11?
We are not sure but could it be that version 11 handles it better ?. I am
attaching the functions on both versions for your review.
Thank you
/*! \brief Send frame to media channel (rtp) */
static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct sip_pvt *p = ast_channel_tech_pvt(ast);
int res = 0;
switch (frame->frametype) {
case AST_FRAME_VOICE:
if
(!(ast_format_cap_iscompatible(ast_channel_nativeformats(ast),
&frame->subclass.format))) {
char s1[512];
ast_log(LOG_WARNING, "Asked to transmit frame type %s,
while native formats is %s read/write = %s/%s\n",
ast_getformatname(&frame->subclass.format),
ast_getformatname_multiple(s1, sizeof(s1),
ast_channel_nativeformats(ast)),
ast_getformatname(ast_channel_readformat(ast)),
ast_getformatname(ast_channel_writeformat(ast)));
return 0;
}
if (p) {
sip_pvt_lock(p);
if (p->t38.state == T38_ENABLED) {
/* drop frame, can't sent VOICE frames while in
T.38 mode */
sip_pvt_unlock(p);
break;
} else if (p->rtp) {
/* If channel is not up, activate early media
session */
if ((ast_channel_state(ast) != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0],
SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING))
{
ast_rtp_instance_update_source(p->rtp);
if (!global_prematuremediafilter) {
p->invitestate =
INV_EARLY_MEDIA;
transmit_provisional_response(p, "183 Session Progress", &p->initreq, TRUE);
ast_set_flag(&p->flags[0],
SIP_PROGRESS_SENT);
}
}
p->lastrtptx = time(NULL);
res = ast_rtp_instance_write(p->rtp, frame);
}
sip_pvt_unlock(p);
}
break;
case AST_FRAME_VIDEO:
if (p) {
sip_pvt_lock(p);
if (p->vrtp) {
/* Activate video early media */
if ((ast_channel_state(ast) != AST_STATE_UP) &&
!ast_test_flag(&p->flags[0],
SIP_PROGRESS_SENT) &&
!ast_test_flag(&p->flags[0], SIP_OUTGOING))
{
p->invitestate = INV_EARLY_MEDIA;
transmit_provisional_response(p, "183
Session Progress", &p->initreq, TRUE);
ast_set_flag(&p->flags[0],
SIP_PROGRESS_SENT);
}
p->lastrtptx = time(NULL);
res = ast_rtp_instance_write(p->vrtp, frame);
}
sip_pvt_unlock(p);
}
break;
case AST_FRAME_TEXT:
if (p) {
sip_pvt_lock(p);
if (p->red) {
ast_rtp_red_buffer(p->trtp, frame);
} else {
if (p->trtp) {
/* Activate text early media */
if ((ast_channel_state(ast) !=
AST_STATE_UP) &&
!ast_test_flag(&p->flags[0],
SIP_PROGRESS_SENT) &&
Re: [asterisk-users] Question about Warning message
thank you, we are using the same configuration files in 13, same setup, just different asterisk version. we just dont see the msgs in the console/logs, it is the same exact voice traffic on both asterisk versions is that something that you set on/off? if that is the case how can it be done? what is the alternative? what are their differences/characteristics? how to choose one over among others? thank you again > From: fbo...@hotmail.com > To: asterisk-users@lists.digium.com > Subject: Question about Warning message > Date: Mon, 23 Feb 2015 12:27:05 -0500 > > > Starting with Asterisk 13.1 we are seeing this WARNING messages a lot in our > logs and console: > > > WARNING[25164][C-0004865e]: chan_sip.c:7364 sip_write: Can't send 10 type > frames with SIP write) > > > We found that line in function "sip_write" inside "chan_sip.c". > > In our previous version (11.2.1) we did not see those messages being printed > (same verbosity level). We compared both versions of the functions and see no > difference at all in the 'default' switch case that handles that. We > think/assume that that function is being called in > different places on each version (11.2-1 vs 13-1). > > We also think it has to do with the asterisk receiving rtp packets with > comfort noise which is not supported by asterisk. > > We would like to know what can we do about it to behave more like the version > 11? > > We are not sure but could it be that version 11 handles it better ?. I am > attaching the functions on both versions for your review. > > Thank you > > > > -- _ -- 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] having trouble to register cisco 7975 with pjsip
Hay guys, got trouble with registration with cisco 7975 Here is the debug : <--- Received SIP request (576 bytes) from UDP:192.168.1.61:49533 ---> REGISTER sip:192.168.1.4 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.61:5060;branch=z9hG4bK35076381 From: ;tag=0c8525a68961001f44d503e2-d9359bd3 To: Call-ID: 0c8525a6-89610004-b972d038-5864c98e@192.168.1.61 Max-Forwards: 70 Date: Tue, 24 Feb 2015 07:13:42 GMT CSeq: 110 REGISTER User-Agent: Cisco-CP7975G/8.5.3 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="437" Supported: (null),X-cisco-xsi-7.0.1 Content-Length: 0 Expires: 3600 <--- Transmitting SIP response (481 bytes) to UDP:192.168.1.61:49531 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.61:5060;rport=49531;received=192.168.1.61;branch=z9hG4bKd16b1eb7 Call-ID: 0c8525a6-89610002-845d0080-f3559596@192.168.1.61 From: ;tag=0c8525a68961001d53245ebc-a1b56549 To: ;tag=z9hG4bKd16b1eb7 CSeq: 110 REGISTER WWW-Authenticate: Digest realm="asterisk",nonce="1424762038/41d5874af9ea9408c257949c309c8aa0",opaque="7f15d8c2312c7b0d",algorithm=md5,qop="auth" Content-Length: 0 username and password are correct, this phone was working with 3CX just fine but won’t work with asterisk for some reason. ( any idea what may cause the problem?-- _ -- 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
