Re: [asterisk-users] packet2packet bridging
Hi, Please clear me on this topic I am confused My log show switching to native rtp. Did this line means that the audio is not coming to the asterisk server any more and asterisk only send the re- invite packet to both the clients ? Am I right or wrong ? On Tue, Jul 8, 2014 at 11:50 PM, Mitul Limbani mi...@enterux.in wrote: No way to avoid bw charges for any of the client if it is behind any sort of NAT. On 08-Jul-2014 8:52 PM, Sameer Rathod sam...@hostnsoft.com wrote: Hi Eric, I am behind nat Is there any solution for the same. My goal is to deduct the balance for the call but free my asterisk server from audio packet load. On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling ewiel...@nyigc.com wrote: I think you will find that direct audio between two endpoints does not work when NAT is involved. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sameer Rathod *Sent:* Tuesday, July 08, 2014 11:18 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] packet2packet bridging Hi Joshua, I had disabled ice support and remover encryption= yes Then also it is showing the same native_rtp in log Could you help me in bypassing asterisk server for audio? please help me I am struggling with it form a long time. On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod sam...@hostnsoft.com wrote: -- Channel SIP/1060-008e left 'native_rtp' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b -- Channel SIP/1061-008f left 'native_rtp' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b == Spawn extension (sameer, 1061, 1) exited non-zero on 'SIP/1060-008e' here are more generated when I cut the call On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod sam...@hostnsoft.com wrote: so In this case If I disable ice support ie commented the icesuppot=yes from all files then also I am getting this output -- Executing [1061@sameer:1] Dial(SIP/1060-008e, SIP/1061) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/1061 -- SIP/1061-008f is ringing -- SIP/1061-008f answered SIP/1060-008e -- Channel SIP/1061-008f joined 'simple_bridge' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b -- Channel SIP/1060-008e joined 'simple_bridge' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from simple_bridge technology to native_rtp 0x7f6800039020 -- Probation passed - setting RTP source address to 192.168.1.176:8000 0x7f6780045810 -- Probation passed - setting RTP source address to 192.168.1.191:8000 On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp jc...@digium.com wrote: Sameer Rathod wrote: yes I had configured icesupport=yes ; Asterisk does not support direct media establishment (with either chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use. -- 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 -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- Regards Sameer Rathod 8109413462 --
Re: [asterisk-users] packet2packet bridging
Put sip debug on to know if reinvite packets are sent. On 09-Jul-2014 1:17 PM, Sameer Rathod sam...@hostnsoft.com wrote: Hi, Please clear me on this topic I am confused My log show switching to native rtp. Did this line means that the audio is not coming to the asterisk server any more and asterisk only send the re- invite packet to both the clients ? Am I right or wrong ? On Tue, Jul 8, 2014 at 11:50 PM, Mitul Limbani mi...@enterux.in wrote: No way to avoid bw charges for any of the client if it is behind any sort of NAT. On 08-Jul-2014 8:52 PM, Sameer Rathod sam...@hostnsoft.com wrote: Hi Eric, I am behind nat Is there any solution for the same. My goal is to deduct the balance for the call but free my asterisk server from audio packet load. On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling ewiel...@nyigc.com wrote: I think you will find that direct audio between two endpoints does not work when NAT is involved. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sameer Rathod *Sent:* Tuesday, July 08, 2014 11:18 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] packet2packet bridging Hi Joshua, I had disabled ice support and remover encryption= yes Then also it is showing the same native_rtp in log Could you help me in bypassing asterisk server for audio? please help me I am struggling with it form a long time. On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod sam...@hostnsoft.com wrote: -- Channel SIP/1060-008e left 'native_rtp' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b -- Channel SIP/1061-008f left 'native_rtp' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b == Spawn extension (sameer, 1061, 1) exited non-zero on 'SIP/1060-008e' here are more generated when I cut the call On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod sam...@hostnsoft.com wrote: so In this case If I disable ice support ie commented the icesuppot=yes from all files then also I am getting this output -- Executing [1061@sameer:1] Dial(SIP/1060-008e, SIP/1061) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/1061 -- SIP/1061-008f is ringing -- SIP/1061-008f answered SIP/1060-008e -- Channel SIP/1061-008f joined 'simple_bridge' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b -- Channel SIP/1060-008e joined 'simple_bridge' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from simple_bridge technology to native_rtp 0x7f6800039020 -- Probation passed - setting RTP source address to 192.168.1.176:8000 0x7f6780045810 -- Probation passed - setting RTP source address to 192.168.1.191:8000 On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp jc...@digium.com wrote: Sameer Rathod wrote: yes I had configured icesupport=yes ; Asterisk does not support direct media establishment (with either chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use. -- 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 -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- Regards Sameer Rathod 8109413462 -- _ -- 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
Re: [asterisk-users] Database and variables
Le 08/07/2014 16:07, Eric Wieling a écrit :
If you are executing database put Agora modele/IVR/AstreinteNagios/1
${ASTR_State} while in the Asterisk CLI, that won't work. You cannot access
DIALPLAN variables from the CLI.
I didn't know that, thanks. Will try another way.
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator
TOOTAI
Sent: Tuesday, July 08, 2014 11:02 AM
To: Asterisk-Users
Subject: [asterisk-users] Database and variables
Hi list,
question regarding the result of a DB query. I have
database put Agora modele/IVR/AstreinteNagios/1 ${ASTR_State}
i=1 and I read the DB with
exten = IVR,n,Set(__PlayMe=${DB(${ASTRSVC}/IVR/${IVR}/${i})})
exten = IVR,n,NoOp(We read ${PlayMe})
Result:
-- Executing [IVR@Automates:8] Set(SIP/laotseu-0001,
__PlayMe=${ASTR_State}) in new stack
-- Executing [IVR@Automates:9] NoOp(SIP/laotseu-0001, Value
of PlayMe is ${ASTR_State}) in new stack
This means that ${ASTR_State} is not considered as a variable but as a
simple alphanumerical chain. What I would like is to display the value
of ASTR_State wich was setted before the PlayMe affectation in the
dialplan. I tried the ${${ASTR_State}} command, no more luck.
Is there a way to archieve what I want to do? A regexp ?
In any dialplan, if you make a NoOp( ${blabla} ) and blabla was not
inizialized, the ${blabla} has an empty value. Why in my case above
${ASTR_State} is not treated as a variable?
Thanks for any suggestion
--
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Re: [asterisk-users] packet2packet bridging
Hi Mitul, I checked that the re-invite packet are sent what I want to check is whether the audio packets is going through the server or not ? On Wed, Jul 9, 2014 at 1:42 PM, Mitul Limbani mi...@enterux.in wrote: Put sip debug on to know if reinvite packets are sent. On 09-Jul-2014 1:17 PM, Sameer Rathod sam...@hostnsoft.com wrote: Hi, Please clear me on this topic I am confused My log show switching to native rtp. Did this line means that the audio is not coming to the asterisk server any more and asterisk only send the re- invite packet to both the clients ? Am I right or wrong ? On Tue, Jul 8, 2014 at 11:50 PM, Mitul Limbani mi...@enterux.in wrote: No way to avoid bw charges for any of the client if it is behind any sort of NAT. On 08-Jul-2014 8:52 PM, Sameer Rathod sam...@hostnsoft.com wrote: Hi Eric, I am behind nat Is there any solution for the same. My goal is to deduct the balance for the call but free my asterisk server from audio packet load. On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling ewiel...@nyigc.com wrote: I think you will find that direct audio between two endpoints does not work when NAT is involved. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sameer Rathod *Sent:* Tuesday, July 08, 2014 11:18 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] packet2packet bridging Hi Joshua, I had disabled ice support and remover encryption= yes Then also it is showing the same native_rtp in log Could you help me in bypassing asterisk server for audio? please help me I am struggling with it form a long time. On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod sam...@hostnsoft.com wrote: -- Channel SIP/1060-008e left 'native_rtp' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b -- Channel SIP/1061-008f left 'native_rtp' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b == Spawn extension (sameer, 1061, 1) exited non-zero on 'SIP/1060-008e' here are more generated when I cut the call On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod sam...@hostnsoft.com wrote: so In this case If I disable ice support ie commented the icesuppot=yes from all files then also I am getting this output -- Executing [1061@sameer:1] Dial(SIP/1060-008e, SIP/1061) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/1061 -- SIP/1061-008f is ringing -- SIP/1061-008f answered SIP/1060-008e -- Channel SIP/1061-008f joined 'simple_bridge' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b -- Channel SIP/1060-008e joined 'simple_bridge' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from simple_bridge technology to native_rtp 0x7f6800039020 -- Probation passed - setting RTP source address to 192.168.1.176:8000 0x7f6780045810 -- Probation passed - setting RTP source address to 192.168.1.191:8000 On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp jc...@digium.com wrote: Sameer Rathod wrote: yes I had configured icesupport=yes ; Asterisk does not support direct media establishment (with either chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use. -- 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 -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- Regards Sameer Rathod 8109413462 -- _ -- 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
Re: [asterisk-users] packet2packet bridging
use tcpdump on the server to see if the RTP traffic is passing through it. On 9 July 2014 10:48, Sameer Rathod sam...@hostnsoft.com wrote: Hi Mitul, I checked that the re-invite packet are sent what I want to check is whether the audio packets is going through the server or not ? On Wed, Jul 9, 2014 at 1:42 PM, Mitul Limbani mi...@enterux.in wrote: Put sip debug on to know if reinvite packets are sent. On 09-Jul-2014 1:17 PM, Sameer Rathod sam...@hostnsoft.com wrote: Hi, Please clear me on this topic I am confused My log show switching to native rtp. Did this line means that the audio is not coming to the asterisk server any more and asterisk only send the re- invite packet to both the clients ? Am I right or wrong ? On Tue, Jul 8, 2014 at 11:50 PM, Mitul Limbani mi...@enterux.in wrote: No way to avoid bw charges for any of the client if it is behind any sort of NAT. On 08-Jul-2014 8:52 PM, Sameer Rathod sam...@hostnsoft.com wrote: Hi Eric, I am behind nat Is there any solution for the same. My goal is to deduct the balance for the call but free my asterisk server from audio packet load. On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling ewiel...@nyigc.com wrote: I think you will find that direct audio between two endpoints does not work when NAT is involved. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sameer Rathod *Sent:* Tuesday, July 08, 2014 11:18 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] packet2packet bridging Hi Joshua, I had disabled ice support and remover encryption= yes Then also it is showing the same native_rtp in log Could you help me in bypassing asterisk server for audio? please help me I am struggling with it form a long time. On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod sam...@hostnsoft.com wrote: -- Channel SIP/1060-008e left 'native_rtp' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b -- Channel SIP/1061-008f left 'native_rtp' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b == Spawn extension (sameer, 1061, 1) exited non-zero on 'SIP/1060-008e' here are more generated when I cut the call On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod sam...@hostnsoft.com wrote: so In this case If I disable ice support ie commented the icesuppot=yes from all files then also I am getting this output -- Executing [1061@sameer:1] Dial(SIP/1060-008e, SIP/1061) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/1061 -- SIP/1061-008f is ringing -- SIP/1061-008f answered SIP/1060-008e -- Channel SIP/1061-008f joined 'simple_bridge' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b -- Channel SIP/1060-008e joined 'simple_bridge' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from simple_bridge technology to native_rtp 0x7f6800039020 -- Probation passed - setting RTP source address to 192.168.1.176:8000 0x7f6780045810 -- Probation passed - setting RTP source address to 192.168.1.191:8000 On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp jc...@digium.com wrote: Sameer Rathod wrote: yes I had configured icesupport=yes ; Asterisk does not support direct media establishment (with either chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use. -- 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 -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- Regards Sameer Rathod 8109413462 -- _ -- 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:
[asterisk-users] switching from simple_bridge technology to native_rtp issue
Hi, with canreinvite=no and directmedia=no I and getting the message in the logs for all calls switching from simple_bridge technology to native_rtp -- Executing [102@mkg:1] Dial(SIP/101-0017, SIP/102) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/102 -- SIP/102-0018 is ringing -- SIP/102-0018 answered SIP/101-0017 -- Channel SIP/101-0017 joined 'simple_bridge' basic-bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab -- Channel SIP/102-0018 joined 'simple_bridge' basic-bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from simple_bridge technology to native_rtp 0x7f427c068a10 -- Probation passed - setting RTP source address to 111.118.250.236:49344 0x7f427c068a10 -- Probation passed - setting RTP source address to 111.118.250.236:49344 0x7f42500168d0 -- Probation passed - setting RTP source address to 111.118.250.236:26326 0x7f42500168d0 -- Probation passed - setting RTP source address to 111.118.250.236:26326 -- Channel SIP/101-0017 left 'native_rtp' basic-bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab -- Channel SIP/102-0018 left 'native_rtp' basic-bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab == Spawn extension (mkg, 102, 1) exited non-zero on 'SIP/101-0017' I cannot understand why asterisk state diff bridges if all works same please can anyone explain me the working bridging concept and how to configure and use bridges to route the rtp externally form asterisk. -- Regards Sameer Rathod 8109413462 -- _ -- 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] packet2packet bridging
Hi Ishfaq, I am getting the the flow as attached Could you please read and check if the rtp is passing directly as I am new and dont know much about this all On Wed, Jul 9, 2014 at 3:24 PM, Ishfaq Malik i...@pack-net.co.uk wrote: use tcpdump on the server to see if the RTP traffic is passing through it. On 9 July 2014 10:48, Sameer Rathod sam...@hostnsoft.com wrote: Hi Mitul, I checked that the re-invite packet are sent what I want to check is whether the audio packets is going through the server or not ? On Wed, Jul 9, 2014 at 1:42 PM, Mitul Limbani mi...@enterux.in wrote: Put sip debug on to know if reinvite packets are sent. On 09-Jul-2014 1:17 PM, Sameer Rathod sam...@hostnsoft.com wrote: Hi, Please clear me on this topic I am confused My log show switching to native rtp. Did this line means that the audio is not coming to the asterisk server any more and asterisk only send the re- invite packet to both the clients ? Am I right or wrong ? On Tue, Jul 8, 2014 at 11:50 PM, Mitul Limbani mi...@enterux.in wrote: No way to avoid bw charges for any of the client if it is behind any sort of NAT. On 08-Jul-2014 8:52 PM, Sameer Rathod sam...@hostnsoft.com wrote: Hi Eric, I am behind nat Is there any solution for the same. My goal is to deduct the balance for the call but free my asterisk server from audio packet load. On Tue, Jul 8, 2014 at 7:51 PM, Eric Wieling ewiel...@nyigc.com wrote: I think you will find that direct audio between two endpoints does not work when NAT is involved. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sameer Rathod *Sent:* Tuesday, July 08, 2014 11:18 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] packet2packet bridging Hi Joshua, I had disabled ice support and remover encryption= yes Then also it is showing the same native_rtp in log Could you help me in bypassing asterisk server for audio? please help me I am struggling with it form a long time. On Wed, Jul 2, 2014 at 8:21 PM, Sameer Rathod sam...@hostnsoft.com wrote: -- Channel SIP/1060-008e left 'native_rtp' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b -- Channel SIP/1061-008f left 'native_rtp' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b == Spawn extension (sameer, 1061, 1) exited non-zero on 'SIP/1060-008e' here are more generated when I cut the call On Wed, Jul 2, 2014 at 8:19 PM, Sameer Rathod sam...@hostnsoft.com wrote: so In this case If I disable ice support ie commented the icesuppot=yes from all files then also I am getting this output -- Executing [1061@sameer:1] Dial(SIP/1060-008e, SIP/1061) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/1061 -- SIP/1061-008f is ringing -- SIP/1061-008f answered SIP/1060-008e -- Channel SIP/1061-008f joined 'simple_bridge' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b -- Channel SIP/1060-008e joined 'simple_bridge' basic-bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b Bridge 3c12ca41-e180-4fc1-80cf-1339b96da42b: switching from simple_bridge technology to native_rtp 0x7f6800039020 -- Probation passed - setting RTP source address to 192.168.1.176:8000 0x7f6780045810 -- Probation passed - setting RTP source address to 192.168.1.191:8000 On Wed, Jul 2, 2014 at 8:13 PM, Joshua Colp jc...@digium.com wrote: Sameer Rathod wrote: yes I had configured icesupport=yes ; Asterisk does not support direct media establishment (with either chan_sip or chan_pjsip) if secure media (SRTP) or ICE is in use. -- 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 -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- Regards Sameer Rathod 8109413462 -- _ -- 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 -- Regards Sameer Rathod 8109413462 -- _ -- Bandwidth and Colocation Provided
Re: [asterisk-users] Asterisk in debian Wheezy 1.8.13.1 vs. Squeeze 1.8.23.1
On Wed, Jul 02, 2014 at 10:05:44PM +0200, Thomas wrote: Hello, in Squeeze Asterisk 1.8.23.1 is installed, Self-installed in Wheezy older version 1.8.13.1~dfsg1-3+deb7u3. From a package. With version 1.8.13.1 I have some problems so I would like to install version 1.8.23.1 used in Squeeze whats running fine for me. How I can do this? Install from source as in Squeeze? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com -- _ -- 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] packet2packet bridging
On Wed, Jul 9, 2014 at 2:47 AM, Sameer Rathod sam...@hostnsoft.com wrote: Hi, Please clear me on this topic I am confused My log show switching to native rtp. Did this line means that the audio is not coming to the asterisk server any more and asterisk only send the re- invite packet to both the clients ? Am I right or wrong ? You are wrong (sorry). All that means is that the bridging has switched to a native RTP bridge. That bridge comes in two variants: a local packet to packet bridge (where the media flows through Asterisk but is not decoded - RTP is merely swapped between ports) and a remote bridge. The remote bridge is where the two channels are in a bridge in Asterisk, but media flows directly between the endpoints. If your endpoints are behind a NAT, then no, you cannot use a remote bridge. No amount of hoping or tinkering will make it so. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://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
Re: [asterisk-users] switching from simple_bridge technology to native_rtp issue
On Wed, Jul 9, 2014 at 4:56 AM, Sameer Rathod sam...@hostnsoft.com wrote:
Hi,
with canreinvite=no and directmedia=no I and getting the message in the logs
for all calls
switching from simple_bridge technology to native_rtp
-- Executing [102@mkg:1] Dial(SIP/101-0017, SIP/102) in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/102
-- SIP/102-0018 is ringing
-- SIP/102-0018 answered SIP/101-0017
-- Channel SIP/101-0017 joined 'simple_bridge' basic-bridge
0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab
-- Channel SIP/102-0018 joined 'simple_bridge' basic-bridge
0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab
Bridge 0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab: switching from
simple_bridge technology to native_rtp
0x7f427c068a10 -- Probation passed - setting RTP source address to
111.118.250.236:49344
0x7f427c068a10 -- Probation passed - setting RTP source address to
111.118.250.236:49344
0x7f42500168d0 -- Probation passed - setting RTP source address to
111.118.250.236:26326
0x7f42500168d0 -- Probation passed - setting RTP source address to
111.118.250.236:26326
-- Channel SIP/101-0017 left 'native_rtp' basic-bridge
0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab
-- Channel SIP/102-0018 left 'native_rtp' basic-bridge
0ad2e3a9-e4be-4b2b-bf55-0357dafcdbab
== Spawn extension (mkg, 102, 1) exited non-zero on 'SIP/101-0017'
I cannot understand why asterisk state diff bridges if all works same
please can anyone explain me the working bridging concept and how to
configure and use bridges to route the rtp externally form asterisk.
I think I just answered this in your other thread, but I'll repeat it here.
First, canreinvite has been deprecated as a naming convention for ...
a long time. It's not even documented any more. The code will accept
it, but all you're doing is setting the directmedia option twice:
} else if (!strcasecmp(v-name, directmedia) ||
!strcasecmp(v-name, canreinvite)) {
ast_set_flag(mask[0], SIP_REINVITE);
ast_clear_flag(flags[0], SIP_REINVITE);
The native RTP bridge in Asterisk 12 manages bridges between two RTP
capable channels. The bridge can either be formed remotely (in which
case the media flows between the endpoints) or locally, in which case
the media is swapped across the ports. It will attempt to perform a
remote bridge if possible, while falling back to a local bridge if a
remote bridge is not possible.
In your particular case, you've explicitly told it to *not* do
directmedia. So it won't perform a remote bridge.
Even if you set directmedia=yes (or one of its variants), you may not
have a successful remote bridge if one of the endpoints is behind a
NAT. The sip.conf sample configuration documentation is actually quite
good on this subject:
;--- MEDIA HANDLING
; By default, Asterisk tries to re-invite media streams to an optimal
path. If there's
; no reason for Asterisk to stay in the media path, the media will be
redirected.
; This does not really work well in the case where Asterisk is outside and the
; clients are on the inside of a NAT. In that case, you want to set
directmedia=nonat.
;
;directmedia=yes; Asterisk by default tries to redirect the
; RTP media stream to go directly from
; the caller to the callee. Some devices do not
; support this (especially if one of
them is behind a NAT).
; The default setting is YES. If you
have all clients
; behind a NAT, or for some other
reason want Asterisk to
; stay in the audio path, you may want
to turn this off.
; This setting also affect direct RTP
; at call setup (a new feature in 1.4
- setting up the
; call directly between the endpoints
instead of sending
; a re-INVITE).
; Additionally this option does not
disable all reINVITE operations.
; It only controls Asterisk generating
reINVITEs for the specific
; purpose of setting up a direct media
path. If a reINVITE is
; needed to switch a media stream to
inactive (when placed on
; hold) or to T.38, it will still be
done, regardless of this
; setting. Note that direct T.38 is
not supported.
Matt
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com http://asterisk.org
--
_
-- Bandwidth and Colocation Provided by
[asterisk-users] How to monitor non-SNMP SIP devices ?
Hi, I'm seeing a trend in which SIP devices such as Yealink SIP phones (with v72 firmware), are dropping support of SNMP in favor of HTTP eventing if may call this as such : when configuring the SIP device, you can define a couple of HTTP URL which triggered when some event occur (end of boot, on hook, ...). How do deal with those devices ? Do you still try to monitor them with usual tools (Nagios, OpenNMS) or do you favor another class of software ? Regards -- _ -- 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 monitor non-SNMP SIP devices ?
asterisk-users-boun...@lists.digium.com wrote on 07/09/2014 10:19:11 AM: From: Olivier oza.4...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com, Date: 07/09/2014 10:19 AM Subject: [asterisk-users] How to monitor non-SNMP SIP devices ? Sent by: asterisk-users-boun...@lists.digium.com Hi, I'm seeing a trend in which SIP devices such as Yealink SIP phones (with v72 firmware), are dropping support of SNMP in favor of HTTP eventing if may call this as such : when configuring the SIP device, you can define a couple of HTTP URL which triggered when some event occur (end of boot, on hook, ...). How do deal with those devices ? Do you still try to monitor them with usual tools (Nagios, OpenNMS) or do you favor another class of software ? We don't monitor our phone endpoints (we do our trunks), but if I were to, I would probably set up a simple webserver with some php that would write the logs to a sql database. What you describe isn't really as good as snmp though, because I can have my monitoring system poll snmp devices, whereas HTTP eventing depends on an event happening to trigger the contact. If the phone goes down hard or locks up, I may not know there is a problem or just no events have happened. I hope at the least, they have a keep alive event that can periodically access the url to indicate all is well. On things I want to monitor, I just don't like the idea of not being able to have my monitoring system talk to them and depending on them talking to my monitoring system. That would probably make me heavily reconsider buying any more of their products if it was something I depended on.-- _ -- 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 monitor non-SNMP SIP devices ?
On 9 July 2014 16:19, Olivier oza.4...@gmail.com wrote: Hi, I'm seeing a trend in which SIP devices such as Yealink SIP phones (with v72 firmware), are dropping support of SNMP in favor of HTTP eventing if may call this as such : when configuring the SIP device, you can define a couple of HTTP URL which triggered when some event occur (end of boot, on hook, ...). How do deal with those devices ? Do you still try to monitor them with usual tools (Nagios, OpenNMS) or do you favor another class of software ? Regards If you set qualify on your peers you could monitor the event stream of the AMI which would show you any end point going unreachable. Regards Ish -- Ishfaq Malik Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: i...@pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, Duplex 2, Ducie House 37 Ducie Street Manchester, M1 2JW COMPANY REG NO. 04920552 -- _ -- 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] PRI congestion instead of busy
I have two servers, each connected to the PTSN via PRI. When I call from site A (951-999-) to site B (555-1212) and the phone at site B is on the phone, I hear the normal ring tone for about 20 seconds, then the message all circuits are busy now. please try your call again latter followed by the congestion tone. Instead, I want this to busy ring and then hang up without any message. Here is a snippet from site A: ... [2014-07-09 09:56:16] VERBOSE[21606][C-dab7] app_dial.c: -- Called DAHDI/g5/5551212 [2014-07-09 09:56:17] VERBOSE[21606][C-dab7] app_dial.c: -- DAHDI/i7/5551212-411b is proceeding passing it to SIP/260-a2f1 [2014-07-09 09:56:17] VERBOSE[21606][C-dab7] app_dial.c: -- DAHDI/i7/5551212-411b is ringing [2014-07-09 09:56:17] VERBOSE[21606][C-dab7] app_dial.c: -- DAHDI/i7/5551212-411b is making progress passing it to SIP/260-a2f1 [2014-07-09 09:56:18] VERBOSE[21606][C-dab7] app_dial.c: -- SIP/260-a2f1 requested media update control 26, passing it to DAHDI/i7/5551212-411b [2014-07-09 09:56:37] VERBOSE[2286][C-dab7] sig_pri.c: -- Span 7: Channel 0/3 got hangup request, cause 16 ... And from site B: ... [2014-07-09 09:56:17] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s@macro-exten-vm:22] GotoIf(DAHDI/i8/951999-59f, 1?s-BUSY,1) in new stack [2014-07-09 09:56:17] VERBOSE[3775][C-0bb1] pbx.c: -- Goto (macro-exten-vm,s-BUSY,1) [2014-07-09 09:56:17] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s-BUSY@macro-exten-vm:1] GotoIf(DAHDI/i8/951999-59f, 0?exit,1) in new stack [2014-07-09 09:56:17] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s-BUSY@macro-exten-vm:2] PlayTones(DAHDI/i8/951999-59f, busy) in new stack [2014-07-09 09:56:17] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s-BUSY@macro-exten-vm:3] Busy(DAHDI/i8/951999-59f, 20) in new stack [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] app_macro.c: == Spawn extension (macro-exten-vm, s-BUSY, 3) exited non-zero on 'DAHDI/i8/951999-59f' in macro 'exten-vm' [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: == Spawn extension (from-did-direct, 803, 2) exited non-zero on 'DAHDI/i8/951999-59f' [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [h@from-did-direct:1] Macro(DAHDI/i8/951999-59f, hangupcall,) in new stack [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf(DAHDI/i8/951999-59f, 1?theend) in new stack [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: -- Goto (macro-hangupcall,s,3) [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf(DAHDI/i8/951999-59f, 0?Set(CDR(recordingfile)=)) in new stack [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup(DAHDI/i8/951999-59f, ) in new stack ... My hunch is that the PRI cause is never set, so site A gets the generic cause 16 (normal call clearing) instead of 17 (user busy). I suspect this is causing site A to get the all circuits are busy now message instead of a busy signal. I thought calling Busy() would cause the PRI cause to get set when used on a channel that is PRI? Should this be manually set instead? Site B details: Asterisk version 11.10.2 Libpri version: 1.4.12 DAHDI version: 2.9.0.1 Freepbx version: 2.11.0.37, distro version 5.211.65-14 -Justin -- _ -- 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] PRI congestion instead of busy
If you use Playtones you should put an Answer and a Wait(1) before the Playtones I recommend using the Hangup app instead. Busy would be Hangup(17). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Wednesday, July 09, 2014 2:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] PRI congestion instead of busy I have two servers, each connected to the PTSN via PRI. When I call from site A (951-999-) to site B (555-1212) and the phone at site B is on the phone, I hear the normal ring tone for about 20 seconds, then the message all circuits are busy now. please try your call again latter followed by the congestion tone. Instead, I want this to busy ring and then hang up without any message. Here is a snippet from site A: ... [2014-07-09 09:56:16] VERBOSE[21606][C-dab7] app_dial.c: -- Called DAHDI/g5/5551212 [2014-07-09 09:56:17] VERBOSE[21606][C-dab7] app_dial.c: -- DAHDI/i7/5551212-411b is proceeding passing it to SIP/260-a2f1 [2014-07-09 09:56:17] VERBOSE[21606][C-dab7] app_dial.c: -- DAHDI/i7/5551212-411b is ringing [2014-07-09 09:56:17] VERBOSE[21606][C-dab7] app_dial.c: -- DAHDI/i7/5551212-411b is making progress passing it to SIP/260-a2f1 [2014-07-09 09:56:18] VERBOSE[21606][C-dab7] app_dial.c: -- SIP/260-a2f1 requested media update control 26, passing it to DAHDI/i7/5551212-411b [2014-07-09 09:56:37] VERBOSE[2286][C-dab7] sig_pri.c: -- Span 7: Channel 0/3 got hangup request, cause 16 ... And from site B: ... [2014-07-09 09:56:17] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s@macro-exten-vm:22] GotoIf(DAHDI/i8/951999-59f, 1?s-BUSY,1) in new stack [2014-07-09 09:56:17] VERBOSE[3775][C-0bb1] pbx.c: -- Goto (macro-exten-vm,s-BUSY,1) [2014-07-09 09:56:17] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s-BUSY@macro-exten-vm:1] GotoIf(DAHDI/i8/951999-59f, 0?exit,1) in new stack [2014-07-09 09:56:17] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s-BUSY@macro-exten-vm:2] PlayTones(DAHDI/i8/951999-59f, busy) in new stack [2014-07-09 09:56:17] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s-BUSY@macro-exten-vm:3] Busy(DAHDI/i8/951999-59f, 20) in new stack [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] app_macro.c: == Spawn extension (macro-exten-vm, s-BUSY, 3) exited non-zero on 'DAHDI/i8/951999-59f' in macro 'exten-vm' [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: == Spawn extension (from-did-direct, 803, 2) exited non-zero on 'DAHDI/i8/951999-59f' [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [h@from-did-direct:1] Macro(DAHDI/i8/951999-59f, hangupcall,) in new stack [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf(DAHDI/i8/951999-59f, 1?theend) in new stack [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: -- Goto (macro-hangupcall,s,3) [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf(DAHDI/i8/951999-59f, 0?Set(CDR(recordingfile)=)) in new stack [2014-07-09 09:56:37] VERBOSE[3775][C-0bb1] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup(DAHDI/i8/951999-59f, ) in new stack ... My hunch is that the PRI cause is never set, so site A gets the generic cause 16 (normal call clearing) instead of 17 (user busy). I suspect this is causing site A to get the all circuits are busy now message instead of a busy signal. I thought calling Busy() would cause the PRI cause to get set when used on a channel that is PRI? Should this be manually set instead? Site B details: Asterisk version 11.10.2 Libpri version: 1.4.12 DAHDI version: 2.9.0.1 Freepbx version: 2.11.0.37, distro version 5.211.65-14 -Justin -- _ -- 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 monitor non-SNMP SIP devices ?
Quoting Ishfaq Malik (i...@pack-net.co.uk): On 9 July 2014 16:19, Olivier oza.4...@gmail.com wrote: Hi, I'm seeing a trend in which SIP devices such as Yealink SIP phones (with v72 firmware), are dropping support of SNMP in favor of HTTP eventing How do deal with those devices ? If you set qualify on your peers you could monitor the event stream of the AMI which would show you any end point going unreachable. This is what i do. Certain 'important' SIP endpoints have a qualify setting in Asterisk and i use AMI (or 'asterisk -rx ...') to query that state with an SNMP-extend hook. HTH. -Sndr. -- | What do sheep count when they want to fall asleep? | 4096R/20CC6CD2 - 6D40 1A20 B9AA 87D4 84C7 FBD6 F3A9 9442 20CC 6CD2 -- _ -- 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] PRI congestion instead of busy
I tried changing the dialplan to use Hangup(17) instead of Playback/Busy. Now instead of ringing for 20 seconds and then getting the all circuits are busy now, I get all circuits are busy now immediately. New snippet from site A: ... [2014-07-09 10:53:43] VERBOSE[23020][C-db0d] app_dial.c: -- Called DAHDI/g5/5551212 [2014-07-09 10:53:43] VERBOSE[23020][C-db0d] app_dial.c: -- DAHDI/i7/5551212-413b is proceeding passing it to SIP/260-a35c [2014-07-09 10:53:44] VERBOSE[23020][C-db0d] app_dial.c: -- DAHDI/i7/5551212-413b is ringing [2014-07-09 10:53:44] VERBOSE[23020][C-db0d] app_dial.c: -- DAHDI/i7/5551212-413b is making progress passing it to SIP/260-a35c [2014-07-09 10:53:51] VERBOSE[23020][C-db0d] app_dial.c: -- DAHDI/i7/5551212-413b answered SIP/260-a35c [2014-07-09 10:53:58] VERBOSE[23020][C-db0d] res_musiconhold.c: -- Started music on hold, class 'default', on DAHDI/i7/5551212-413b [2014-07-09 10:55:06] VERBOSE[23020][C-db0d] res_musiconhold.c: -- Stopped music on hold on DAHDI/i7/5551212-413b [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: -- Executing [h@macro-dialout-trunk:1] Macro(SIP/260-a35c, hangupcall,) in new stack [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf(SIP/260-a35c, 1?theend) in new stack [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: -- Goto (macro-hangupcall,s,3) [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf(SIP/260-a35c, 0?Set(CDR(recordingfile)=)) in new stack [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup(SIP/260-a35c, ) in new stack [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/260-a35c' in macro 'hangupcall' [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/260-a35c' [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] chan_dahdi.c: -- Hungup 'DAHDI/i7/5551212-413b' [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/260-a35c' in macro 'dialout-trunk' [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: == Spawn extension (from-internal, 5551212, 5) exited non-zero on 'SIP/260-a35c' ... New snippet from site B: ... [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Goto (macro-exten-vm,s-BUSY,1) [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [s-BUSY@macro-exten-vm:1] GotoIf(DAHDI/i8/951999-603, 0?exit,1) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [s-BUSY@macro-exten-vm:2] Hangup(DAHDI/i8/951999-603, 17) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] app_macro.c: == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'DAHDI/i8/951999-603' in macro 'exten-vm' [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: == Spawn extension (from-did-direct, 803, 2) exited non-zero on 'DAHDI/i8/951999-603' [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [h@from-did-direct:1] Macro(DAHDI/i8/951999-603, hangupcall,) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf(DAHDI/i8/951999-603, 1?theend) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Goto (macro-hangupcall,s,3) [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf(DAHDI/i8/951999-603, 0?Set(CDR(recordingfile)=)) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup(DAHDI/i8/951999-603, ) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'DAHDI/i8/951999-603' in macro 'hangupcall' [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: == Spawn extension (from-did-direct, h, 1) exited non-zero on 'DAHDI/i8/951999-603' [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] chan_dahdi.c: -- Hungup 'DAHDI/i8/951999-603' ... It seems that now the PRI cause isn't getting set at all. -Justin From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, July 09, 2014 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI congestion instead of busy If you use Playtones you should put an Answer and a Wait(1) before the Playtones I recommend using the Hangup app instead. Busy would be Hangup(17). From: asterisk-users-boun...@lists.digium.com
[asterisk-users] Pickup problem
I found a very strange proble whit two asterisk servers in the same network. Scenario Asterisk A with extensions 5XX Asterisk B with extensions 2XX There is NO link between the two asterisks. Call from 501 to 503, 503 ringing Call from 201 to 203, 203 ringing The 202 extension comand a pickup (i dont manage this Asterisk, i think with the Pickup command). The 202 answer the 501 call and not the 201. extensions 5XX are all SNOM extensions 2XX are all Grandstream GX2000 This is the first time i can see two asterisk in the same net so... Why? :-) Thnks. -- _ -- 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] PRI congestion instead of busy
I figured it out - the busy tone was being generated by the local end. It seems that upon receiving a hangup, freepbx tries the next trunk. In this case there isn't any other trunk, so I get all circuits are busy now. For anyone interested, the site B playback()/busy() condition has been submitted as bug# 7706 - http://issues.freepbx.org/browse/FREEPBX-7706 The site A truck failover retry on hangupcause's that shouldn't be retried has been submitted as bug# 7705 - http://issues.freepbx.org/browse/FREEPBX-7705 -Justin From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Wednesday, July 09, 2014 11:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] PRI congestion instead of busy I tried changing the dialplan to use Hangup(17) instead of Playback/Busy. Now instead of ringing for 20 seconds and then getting the all circuits are busy now, I get all circuits are busy now immediately. New snippet from site A: ... [2014-07-09 10:53:43] VERBOSE[23020][C-db0d] app_dial.c: -- Called DAHDI/g5/5551212 [2014-07-09 10:53:43] VERBOSE[23020][C-db0d] app_dial.c: -- DAHDI/i7/5551212-413b is proceeding passing it to SIP/260-a35c [2014-07-09 10:53:44] VERBOSE[23020][C-db0d] app_dial.c: -- DAHDI/i7/5551212-413b is ringing [2014-07-09 10:53:44] VERBOSE[23020][C-db0d] app_dial.c: -- DAHDI/i7/5551212-413b is making progress passing it to SIP/260-a35c [2014-07-09 10:53:51] VERBOSE[23020][C-db0d] app_dial.c: -- DAHDI/i7/5551212-413b answered SIP/260-a35c [2014-07-09 10:53:58] VERBOSE[23020][C-db0d] res_musiconhold.c: -- Started music on hold, class 'default', on DAHDI/i7/5551212-413b [2014-07-09 10:55:06] VERBOSE[23020][C-db0d] res_musiconhold.c: -- Stopped music on hold on DAHDI/i7/5551212-413b [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: -- Executing [h@macro-dialout-trunk:1] Macro(SIP/260-a35c, hangupcall,) in new stack [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf(SIP/260-a35c, 1?theend) in new stack [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: -- Goto (macro-hangupcall,s,3) [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf(SIP/260-a35c, 0?Set(CDR(recordingfile)=)) in new stack [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup(SIP/260-a35c, ) in new stack [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/260-a35c' in macro 'hangupcall' [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/260-a35c' [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] chan_dahdi.c: -- Hungup 'DAHDI/i7/5551212-413b' [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] app_macro.c: == Spawn extension (macro-dialout-trunk, s, 22) exited non-zero on 'SIP/260-a35c' in macro 'dialout-trunk' [2014-07-09 10:55:07] VERBOSE[23020][C-db0d] pbx.c: == Spawn extension (from-internal, 5551212, 5) exited non-zero on 'SIP/260-a35c' ... New snippet from site B: ... [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Goto (macro-exten-vm,s-BUSY,1) [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [s-BUSY@macro-exten-vm:1] GotoIf(DAHDI/i8/951999-603, 0?exit,1) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [s-BUSY@macro-exten-vm:2] Hangup(DAHDI/i8/951999-603, 17) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] app_macro.c: == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'DAHDI/i8/951999-603' in macro 'exten-vm' [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: == Spawn extension (from-did-direct, 803, 2) exited non-zero on 'DAHDI/i8/951999-603' [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [h@from-did-direct:1] Macro(DAHDI/i8/951999-603, hangupcall,) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [s@macro-hangupcall:1] GotoIf(DAHDI/i8/951999-603, 1?theend) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Goto (macro-hangupcall,s,3) [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [s@macro-hangupcall:3] ExecIf(DAHDI/i8/951999-603, 0?Set(CDR(recordingfile)=)) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] pbx.c: -- Executing [s@macro-hangupcall:4] Hangup(DAHDI/i8/951999-603, ) in new stack [2014-07-09 10:47:03] VERBOSE[7133][C-0c69] app_macro.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'DAHDI/i8/951999-603' in macro
[asterisk-users] How to know if the current call has been answer()'ed
Is there a channel variable / status indicator / function that indicates if the current channel has been answer()'ed? -Justin -- _ -- 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 know if the current call has been answer()'ed
Justin Killen wrote:
Is there a channel variable / status indicator / function that indicates
if the current channel has been answer()’ed?
${CHANNEL(state)} will return the state the channel is currently in. If
the channel is answered the state will be Up.
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
--
_
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[asterisk-users] busy() not setting PRI_CAUSE
Okay, I think I need a sanity check here - If I call a person that's on the phone, I should get a busy signal. Now more specifically, a call comes into the pbx via PRI. The destination dialplan runs busy(20). Now, the PRI causecode should get set to 17 (user busy) so that the originating end can play a busy tone, correct? -Justin -- _ -- 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] busy() not setting PRI_CAUSE
Generally if you want to send a cause 17 to the caller you would use Hangup(17) and let the caller's switch generate the busy tone. If the dialplan has already answered the call, then you might want to use Busy or Playtones. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Wednesday, July 09, 2014 8:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] busy() not setting PRI_CAUSE Okay, I think I need a sanity check here - If I call a person that's on the phone, I should get a busy signal. Now more specifically, a call comes into the pbx via PRI. The destination dialplan runs busy(20). Now, the PRI causecode should get set to 17 (user busy) so that the originating end can play a busy tone, correct? -Justin -- _ -- 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] busy() not setting PRI_CAUSE
The description of busy() in the asterisk documentation wiki states: This application will indicate the busy condition to the calling channel. Wouldn't 'indicate the busy condition' on a PRI channel imply setting cause 17? -Justin From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, July 09, 2014 4:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] busy() not setting PRI_CAUSE Generally if you want to send a cause 17 to the caller you would use Hangup(17) and let the caller's switch generate the busy tone. If the dialplan has already answered the call, then you might want to use Busy or Playtones. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Killen Sent: Wednesday, July 09, 2014 8:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] busy() not setting PRI_CAUSE Okay, I think I need a sanity check here - If I call a person that's on the phone, I should get a busy signal. Now more specifically, a call comes into the pbx via PRI. The destination dialplan runs busy(20). Now, the PRI causecode should get set to 17 (user busy) so that the originating end can play a busy tone, correct? -Justin -- _ -- 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] PJSIP Transfer not working
I tried to do what I with regular SIP to Transfer a call via 302 Redirect. In asterisk 12 we need to add the Tech, or not, but in any case, there is no transfer done. The call is closed. Here is a trace. How do I do this? [Jul 9 21:39:29] DEBUG[47716][C-0002]: pbx.c:4869 pbx_extension_helper: Launching 'Transfer' -- Executing [17274428141@redirect:30] Transfer(PJSIP/Client.1.1.1.1-0002, PJSIP/17274428141;rn=+1813402;npdi@1.1.1.1) in new stack [Jul 9 21:39:29] DEBUG[47716][C-0002]: pbx.c:4869 pbx_extension_helper: Launching 'Verbose' -- Executing [17274428141@redirect:31] Verbose(PJSIP/Client.1.1.1.1-0002, 2,Transferred: 17274428141;rn=+1813402;npdi@1.1.1.1) in new stack == Transferred: 17274428141;rn=+1813402;npdi@1.1.1.1 -- Auto fallthrough, channel 'PJSIP/Client.1.1.1.1-0002' status is 'UNKNOWN' [Jul 9 21:39:29] DEBUG[47716][C-0002]: channel.c:2597 ast_softhangup_nolock: Soft-Hanging (0x10) up channel 'PJSIP/Client.1.1.1.1-0002' [Jul 9 21:39:29] DEBUG[47716][C-0002]: channel.c:2753 ast_hangup: Hanging up channel 'PJSIP/Client.1.1.1.1-0002' [Jul 9 21:39:29] DEBUG[47716][C-0002]: chan_pjsip.c:1578 hangup_cause2sip: AST hangup cause 0 (no match found in PJSIP) --- Transmitting SIP response (369 bytes) to UDP:1.1.1.1:49260 --- SIP/2.0 603 Decline v: SIP/2.0/UDP 1.1.1.1:49260;rport;received=1.1.1.1;branch=z9hG4bK-d8754z-22994e127365d474-1---d8754z- i: MmFjNDM4NDc2NmFhZWNiYTU2MDQ1YmNjNGVmYmMyOTY f: 957408 sip:957408@8.26.191.189;tag=82c82c1d t: sip:17274428141@8.26.191.189;tag=09f3a67a-f457-46d1-8d16-243478ac3859 CSeq: 1 INVITE Reason: Q.850;cause=0 l: 0 -- _ -- 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
