[asterisk-users] Remote UNIX Connection Hanging Asterisk
Hi friends, I am facing a problem with my asterisk 1.2 PBX. The problem is because of the CLI message Remote UNIX Connection. After 2 days of a server reboot, this message starts coming. After it starts coming it still works well for few more hours, but then the asterisk hangs. During this time calls are still landing on the system, and calls are going forward upto the queue. Once the calls are placed in the queue, it never hits on the agents extensions. I am using AgentCallBackLogin and there are free agents available. But call never hits the agents. Once it happens, a service asterisk restart solves the problem. But it again comes after few hours again. Kindly help me to solve this problem. Thanks Regards Shanavaz. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question regarding SIP 183 Session Progress handling in Asterisk
Dear Asterisk community! I am having trouble with a project concerning the 183 Session Progress SIP messages. Asterisk seems to only accept these when there is also a Session Description (SDP) included in the message. I also verified this by looking at the code. However for a project we are working with a trunk to a third party system (Alcatel) and they are insisting that this behavior is non-compliant with RFC3261 (SIP). So can someone please tell me the reason, why Asterisk does not support 183 messages without SDP as this would really help me finding arguments in this situation. So far Alcatel just tells us that this is not SIP-compliant and that we have to change things on the Asterisk side, but I'm not quite sure that this is really the case and having arguments could help me clarifying this situation. Thanks in advance. With best regards Florian Floimair Technical Support COMMEND INTERNATIONAL GMBH A-5020 Salzburg, Saalachstraße 51 Tel: +43-662-85 62 25 312 Sip: f.floim...@commend.com file:///T:\KAT\Signaturen\=%22sip:f.floim...@commend.com%22 Fax: +43-662-85 62 26 f.floim...@commend.com mailto:f.floim...@commend.com http://www.commend.com http://www.commend.com/ Security and Communication by Commend FN 178618z | LG Salzburg ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
On 27 Jun 2009, at 10:06, Olivier wrote: Hi, As many remember, almost one year this Skype for Asterisk extension program was announced. Has anyone tried it ? Is there any available pricelist ? I've just had a talk on Skype for Asterisk accepted at Astricon (www.astricon.net ), so if you can wait that long, you come along and I'll try and tell you what SFA can do. In the meanwhile - it often crops up on the voipusers conference (www.vuc.me ) on a Friday. In fact I've been running an experiment allowing people to call the conference from Skype (using SFA of course). Feel free to call in and try it this Friday. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo and static on PRI with errors.
Hi there, I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. A little background: Server: HP DL145 G3 Dual Opteron 246 with 2GB ram and a brand new OpenVox D110P http://www.voipon.co.uk/openvox-d110p-pci-isdn-pri-card-p-658.html Dual 250GB SATA disks in Software Raid 1 Running Ubuntu 9.04 Jaunty, but I had the same problems on Intrepid and Hardy. Versions: Asterisk: 1.4.2, 1.6.0.10, 1.6.2beta4 libpri: 1.4.10 dahdi: 2.2.0-current Asterisk works fine for SIP calls, as long as they don't touch the outside world via the PRI card. This pastebin contains the console log from asterisk -vcg http://pastebin.com/f780c591e There are lots of chan_dahdi errors. Occasionally, it claims to run out of channels and terminates the active calls. This is the contents of /etc/dahdi/system.conf http://pastebin.com/f1f654235 This is the contents of /etc/asterisk/chan_dahdi.conf http://pastebin.com/f7ef35e72 This is /proc/interrupts http://pastebin.com/f61cd8398 This is lsmod http://pastebin.com/m56105bf5 I've tried stuff like binding the processor affinity of the modules to one or the other processor, I've tried changing the slot the card is in. I asked similar questions on #asterisk, and tried their suggestions. Nothing seems to work. Any help would be graciously recieved. I'm pretty much all out of ideas. Tom -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I add one h323 endpoint to register at asterisk?
Hello Can I configure one h323 endpoint to register to asterisk? Which asterisk version can support this? Regards Bilal ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Remote UNIX Connection Hanging Asterisk
On Tue, Jun 30, 2009 at 10:38:25AM +0400, Shanavaz E A wrote: Hi friends, I am facing a problem with my asterisk 1.2 PBX. What version, exactly? The problem is because of the CLI message Remote UNIX Connection. Read: 'asterisk -r' connecting. After 2 days of a server reboot, this message starts coming. After it starts coming it still works well for few more hours, but then the asterisk hangs. I suppose your problem is that Asteirsk hangs. How exactly do you see that it hangs? During this time calls are still landing on the system, and calls are going forward upto the queue. Once the calls are placed in the queue, it never hits on the agents extensions. I am using AgentCallBackLogin and there are free agents available. But call never hits the agents. Once it happens, a service asterisk restart solves the problem. But it again comes after few hours again. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial Chan_local Usage
Hello! I am trying to set up a dialplan that uses the Local channel type: [default] exten = s,1,dial(local/2...@dialplan/n) [dailplan] exten = 220,1,saydigits(123) exten = 220,2,dial(SIP/120||m) The calling party does not hear any of the digits nor the music on hold. What should be done so the sound is sent back to the original call? Thank you for your help, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Opensips+asterisk problem
Hi all
After a long iam back to forum
back to my own topic and several readings done on this forum
how people doing same kind of setup what iam trying to achive
so here i have done some good developements
for testing iam doing all in one Server
Step1 :
Installed in Fresh BOX with Debian
Asterisk and A2B working Fine
Step2 : registered with SIP account iam able to make calls successfully
Step3 :
installed Opensips
Made Subscribers to view from A2b Database
Step4 : changed Asterisk port from 5060 to 5062
Step5 : Opensip config made changes to register users with Opensips
and when they dial 001X call send to Asterisk box
route[3]{
if (uri =~ sip:001[0...@*){
log(1, Forwarding to Asterisk \n);
rewritehostport(A2b-asterisk-IP:5062);
route(1);
exit;
}
Works Fine, No problems as of now
But to go in advance, i want to use Number of * boxes to achive more Load
Step5 : added Dispatcher Module in the Opensips
loadmodule dispatcher.so
.
.
.
modparam(dispatcher,list_file,/usr/local/etc/opensips/dispatcher.cfg)
.
.
.
.
changed route to use dispatcher
route[3]{
if (uri =~ sip:001[0...@*){
log(1, Forwarding to Asterisk \n);
ds_select_dst(2,4);
forward();
route(1);
exit;
}
My dispatcher Config Looks like below
dispatcher.cfg
2 sip:a2b-asterisk-ip:5062
2 sip:a2b-asterisk-ip2:5062
I have restarted Opensips
when i dial 0017XX number the call send Opensips to Asterisk
Jun 30 01:12:28 opensips[25868]: Forwarding to Asterisk
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: set [2]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: alg hash [1]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: selected
[4-2/1] sip:a2b-asterisk-ip:5062
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:mk_proxy: doing DNS
lookup...
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:forward_request:
sending:#012INVITE sip:0017x...@opensips-ip:5060
SIP/2.0#015#012Record-Route: sip:opensips-ip;lr=on#015#012Via: SIP/2.0/UDP
opensips-ip;branch=z9hG4bK28178282572929210914#015#012Via: SIP/2.0/UDP
ip-phone-ip:5060;received=ip-phone-ip;branch=z9hG4bK28178282572929210914;rport=5060#015#012From:
4720779942 sip:4720779...@opensips-ip:5060;tag=1966722825#015#012To:
0017325824631 sip:0017...@opensips-ip:5060#015#012Call-ID:
32167199575863-11502744529...@ip-phoneip#015#012cseq: 2
INVITE#015#012Contact:
sip:4720779...@ipphone-ip:5060#015#012Proxy-Authorization:
Digest username=4720779942, realm=asterisk, nonce=79ee65ba,
uri=sip:0017xxx...@opensips-ip:5060,
response=3e182f165a5663d0b145d6b55d34e94b,
algorithm=MD5#015#012Max-Forwards: 69#015#012Supported:
replaces#015#012User-Agent: Voip Phone 1.0#015#012Allow: INVITE, ACK,
OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK,
UPDATE#015#012Content-Type: application/sdp#015#012Content-Length:
319#015#012#015#012v=0#015#012o=4720779942 17025328 32005127 IN IP4
202.63.111.2#015#012s=A conversation#015#012c=IN IP4 ip-phone-ip#015#012t=0
0#015#012m=audio 10028 RTP/AVP 18 4 8 0 9 101#015#012a=rtpmap:18
G729/8000#015#012a=rtpmap:4 G723/8000#015#012a=rtpmap:8
PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:9
G722/16000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=sendrecv#015#012.
opensips[25868]: DBG:core:forward_request: orig. len=1087, new_len=1220,
proto=1
when i ngrep
U 2009/06/30 01:59:20.770599 ipphone:5060 - asterisk-a2b-ip:5060
INVITE sip:0017x...@asterisk-a2b-ip:5060 SIP/2.0.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport.
From: 4720779942 sip:4720779...@asterisk-a2b-ip:5060;tag=3037030266.
To: 0017 sip:0017x...@asterisk-a2b-ip:5060.
Call-ID: 14399316162240-7371067914...@ipphone.
CSeq: 2 INVITE.
Contact: sip:4720779...@ipphone:5060.
Proxy-Authorization: Digest username=4720779942, realm=asterisk,
nonce=07ba8624, uri=sip:0017x...@asterisk-a2b-ip:5060,
response=5dbe9b2937d0bc3f6e8d25052fff0b6a, algorithm=MD5.
Max-Forwards: 70.
Supported: replaces.
User-Agent: Voip Phone 1.0.
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE,
PRACK, UPDATE.
Content-Type: application/sdp.
Content-Length: 319.
.
v=0.
o=4720779942 69102627 18481147 IN IP4 ipphone.
s=A conversation.
c=IN IP4 ipphone.
t=0 0.
m=audio 10034 RTP/AVP 18 4 8 0 9 101.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:9 G722/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
U 2009/06/30 01:59:20.774528 asterisk-a2b-ip:5060 - ipphone:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport=5060.
From: 4720779942 sip:4720779...@asterisk-a2b-ip:5060;tag=3037030266.
To: 0017 sip:0017x...@asterisk-a2b-ip:5060.
Call-ID: 14399316162240-7371067914...@ipphone.
CSeq: 2 INVITE.
Server: OpenSIPS (1.5.1-notls (i386/linux)).
Content-Length: 0.
.
U 2009/06/30 01:59:21.650498 asterisk-a2b-ip:5060 - ipphone:5060
SIP/2.0 407 Proxy
Re: [asterisk-users] Dial Chan_local Usage
Hello, Oddly enough, sound is sent to original caller if it is a registered SIP device on the server. If the caller is remote, than nothing is passed back. Any help will be greatly appreciated, Elliot On Tue, Jun 30, 2009 at 12:13 PM, Elliot Murdockmurdo...@gmail.com wrote: Hello! I am trying to set up a dialplan that uses the Local channel type: [default] exten = s,1,dial(local/2...@dialplan/n) [dailplan] exten = 220,1,saydigits(123) exten = 220,2,dial(SIP/120||m) The calling party does not hear any of the digits nor the music on hold. What should be done so the sound is sent back to the original call? Thank you for your help, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Issue (1.4.21.1)
What version do you mean.. 1.6? Upgrading might be a option, but we cant loose any functionality/stability - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 14:57:26 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) I think the handling of this may have improved in later versions of Asterisk - is an upgrade an option? (I tested this with a newer version of Asterisk recently, and it behaved how you were hoping it would behave) PaulH Kev Szaszvari wrote: The strange thing is, Queue calls are working as per expected. If they get a call from the queue they wont get another until the 1st call is done. Its only when the agent received a direct call or a internal call from another staff member, the queue continues to ring their phone. - Original Message - From: Kev Szaszvari [mailto:k...@mailcall.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:36:32 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) It appears that that option is set from queues.conf [ops] musicclass = default strategy = leastrecent timeout = 5 retry = 1 wrapuptime= 3 autofill = yes autopause = no maxlen = 0 joinempty = yes leavewhenempty = no ringinuse = no - Original Message - From: Paul Hales [mailto:pdha...@optusnet.com.au] To: Asterisk Users Mailing List - Non-Commercial Discussion [mailto:asterisk-us...@lists.digium.com] Sent: Tue, 30 Jun 2009 11:01:40 +1000 Subject: Re: [asterisk-users] Queue Issue (1.4.21.1) The queue option ringinuse = no might be what you are looking for. PaulH Kev Szaszvari wrote: Hi All I am using asterisk 1.4.21.1 Im not sure if this is a issue but it has become one for me :) When agents are logged in to a queue (AgentCallBackLogin) and they receive a direct line call or a transfer they still receive queue calls. EG Someone in our company transfers a call to a agent - When on the transferred call the queue is still trying to ring the agents phone. I tried setting call-limit = 1 but then the agents lost the ability to announce transfer. Has anyone solved this before? Kev This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is privileged, confidential or copyright. You are hereby notified that any dissemination, distribution or copying of this communication is strictly prohibited without the authority of the sender. If you have received this e-mail message in error or are not the intended recipient, please delete and destroy all copies and notify us immediately by return mail. Any views expressed in this communication are those of the individual sender, except where the sender specifically states otherwise. If you no longer want to receive notifications, simply reply to this e-mail. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This Communication is intended only for the use of the individual or entity to which it is addressed and may contain information that is
[asterisk-users] DUNDi Errors (ENCREJ)
Hello users.
i am planning to implement the dundi protocol among 3 servers
where the real channels residing in 2 servers and the remaining one
is only for routing purpose..
here is how my config files
#Routing_server
routing server -192.168.1.11
node1-192.168.1.21
node2-192.168.1.31
i)dundi.conf
dundi=dundicontext,0,IAX2,priv:${secr...@192.168.1.11/${NUMBER},nopartial
[MACaddress of node1]
model=symmetric
host = 192.168.1.21
inkey = priv
outkey = priv
include = priv
permit = priv
qualify = yes
order=primary
;[MAC oF system node2];
;model=symmetric
;host = 192.168.1.31
;inkey = priv
;outkey = priv
;include = priv
;permit = priv
;qualify = yes
;order=secondary
2)extension.conf
[dundicontext]
include = lookupdundi
[lookupdundi]
switch = DUNDi/dundi
3)iax.conf
[priv]
dbsecret=dundi/secret
type=friend
context=dundicontext
- when i tested the dundi show peers in my server the 2 nodes
information i was able to see
- when i used dundi lookup 2...@dundi
i am getting this error
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response)
Flags: 00 STrans: 16791 DTrans: 30106 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 30106 DTrans: 16791 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: ENCRYPT (Command)
Flags: 00 STrans: 26692 DTrans: 0 [192.168.1.11:4520]
ENTITY IDENT: 00:23:7d:93:f7:5e
KEYCRC32: 4234245369
ENCDATA : [IV 11dc622321b7c86ae1e7016c4fe4101d] 4 encrypted blocks
Tx-Frame Retry[No] -- OSeqno: 000 ISeqno: 001 Type: ENCREJ (Response)
Flags: 00 STrans: 22476 DTrans: 26692 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 001 ISeqno: 001 Type: ACK (Response)
Flags: 00 STrans: 26692 DTrans: 22476 [192.168.1.11:4520] (Final)
Rx-Frame Retry[No] -- OSeqno: 000 ISeqno: 000 Type: ENCRYPT (Command)
Flags: 00 STrans: 00299 DTrans: 0 [192.168.1.11:4520]
ENTITY IDENT: 00:23:7d:93:f7:5e
SHAREDKEY : [ 66 df 0a c5 75 59 2f 75 bc 48 bd c8 39 6c 33 df
73 37 85 10 86 ed b5 da 4c 88 a7 c5 00 f0 ab d0 2a 9b b3 71 86 7a c4
53 dd dc 4f 29 fb 43 b4 17 d9 91 e1 df 5b 6f 6d c2 b0 f7 d2 f9 f0 b8
3b 0c 0e 7d af ef 8e 4f cf 9f 7e ca 50 b2 04 97 60 2b cb df fd 97 82
d4 bf a0 cf 9a 66 60 11 19 bc 6b 63 30 a5 05 2f 9e a7 63 1d 90 f6 ac
13 23 39 30 33 1d 29 7a 0 6 da 52 5d b0 d7 e7 3f e7 ef 2d a1 ]
SIGNATURE : [ 65 da f2 7a 0f e1 ea 40 73 56 bc 78 d0 05 c0 c3
ec 4c 97 53 cc 2f 2f 97 01 1a 0d ee 8f 21 8f 5a c2 65 91 6d 32 16 dc
27 75 f6 12 9f 3e f3 bd 34 29 9e c9 af 8d 03 ef 43 7c f4 4d 48 e6 cc
70 af 86 89 ef 24 78 3e c3 71 be cb 55 2c e3 79 19 61 2b 34 d4 8f 62
f6 99 8d 27 9f af 56 a3 8b 30 c6 a1 42 de e5 92 4b f0 8f a2 90 91 86
27 fd 0f 7f 1d b6 4a f7 7 2 53 95 d9 d2 14 03 c6 fd b9 9e 5a ]
ENCDATA : [IV 11dc622321b7c86ae1e7016c4fe4101d] 4 encrypted blocks
- To resolve this i tried to remove all keys in all servers and once
again created and
distributed the loaded in each system with keys init command but
stilll i am
getting the same error
can anybody help me out???
Thanks and regards
srinivas antarvedi
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Re: [asterisk-users] Dial Chan_local Usage
Hello! I needed to answer the local call for any sound to pass through: [default] exten = s,1,dial(local/2...@dialplan/n) [dailplan] exten = 220,1,answer() exten = 220,2,saydigits(123) exten = 220,3,dial(SIP/120||m) From my understanding, the answer command only answers the local call, but the final dial at priority 3 will remain unanswered. Thanks, Elliot On Tue, Jun 30, 2009 at 12:31 PM, Elliot Murdockmurdo...@gmail.com wrote: Hello, Oddly enough, sound is sent to original caller if it is a registered SIP device on the server. If the caller is remote, than nothing is passed back. Any help will be greatly appreciated, Elliot On Tue, Jun 30, 2009 at 12:13 PM, Elliot Murdockmurdo...@gmail.com wrote: Hello! I am trying to set up a dialplan that uses the Local channel type: [default] exten = s,1,dial(local/2...@dialplan/n) [dailplan] exten = 220,1,saydigits(123) exten = 220,2,dial(SIP/120||m) The calling party does not hear any of the digits nor the music on hold. What should be done so the sound is sent back to the original call? Thank you for your help, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Echo and static on PRI with errors
Hi there, I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. A little background: Server: HP DL145 G3 Dual Opteron 246 with 2GB ram and a brand new OpenVox D110P http://www.voipon.co.uk/openvox-d110p-pci-isdn-pri-card-p-658.html Dual 250GB SATA disks in Software Raid 1 Running Ubuntu 9.04 Jaunty, but I had the same problems on Intrepid and Hardy. Versions: Asterisk: 1.4.2, 1.6.0.10, 1.6.2beta4 libpri: 1.4.10 dahdi: 2.2.0-current Asterisk works fine for SIP calls, as long as they don't touch the outside world via the PRI card. This pastebin contains the console log from asterisk -vcg http://pastebin.com/f780c591e There are lots of chan_dahdi errors. Occasionally, it claims to run out of channels and terminates the active calls. This is the contents of /etc/dahdi/system.conf http://pastebin.com/f1f654235 This is the contents of /etc/asterisk/chan_dahdi.conf http://pastebin.com/f7ef35e72 This is /proc/interrupts http://pastebin.com/f61cd8398 This is lsmod http://pastebin.com/m56105bf5 I've tried stuff like binding the processor affinity of the modules to one or the other processor, I've tried changing the slot the card is in. I asked similar questions on #asterisk, and tried their suggestions. Nothing seems to work. Any help would be graciously recieved. I'm pretty much all out of ideas. Tom -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Redirect with ExtraChannel on Bridged call give AMI event with second channel name AsyncGoto/...ZOMBIE
Originally posted on asterisk-dev with no response for 5 days, so posting it to the wider audience now. Asterisk Release 1.6.1.1 Scenario:- 1. 2 SIP peers (Zoiper softphone, if it matters) registered as 901 and 902 2. Using AMI, 901 is Originated 3. When 901 answers, it is Redirected to an extension exten = dial,1,Dial(SIP/902) 4. 902 rings, then answers 5. AMI recieves the channel events for 902, followed by Bridge event 1. Event: Bridge Privilege: call,all Bridgestate: Link Bridgetype: core Channel1: SIP/901-007f0e98 Channel2: SIP/902-007fe948 Uniqueid1: 1246031137.3 Uniqueid2: 1246031140.4 CallerID1: NODID CallerID2: dial 6. 901 and 902 are perfectly bridged and can talk 7. Now after some time, using AMI, both channels are Redirected to an extension exten = calllegwait,1,Wait(60) 8. AMI recieves the event:- Event: Unlink Privilege: call,all Channel1: SIP/901-007f0e98 Channel2: AsyncGoto/SIP/902-007fe948ZOMBIE Uniqueid1: 1246031137.3 Uniqueid2: 1246031140.4 CallerID1: NODID CallerID2: (null) 2 Issues here:- 1. Why is the Channel2: AsyncGoto/SIP/902-007fe948ZOMBIE instead of just SIP/902-007fe948 2. Why isn't there a Bridge event (with, ofcource, Bridgestate: Unlink) Log snippets below:- *Dial application being launched* [Jun 26 22:24:14] DEBUG[3668]: pbx.c:3179 pbx_extension_helper: Launching 'Dial' -- Executing [d...@from-manager-core:1] Dial(SIP/901-007f0e98, SIP/902,6,6) in new stack *902 answers* [Jun 26 22:24:15] DEBUG[11643]: chan_sip.c:10862 build_route: build_route: Contact hop: sip:9...@10.10.1.162:5060 ;rinstance=9e5f63e47063d77c;transport=UDP [Jun 26 22:24:15] DEBUG[11643]: chan_sip.c:2872 __sip_xmit: Trying to put 'ACK sip:90' onto UDP socket destined for 10.10.1.162:5060 -- SIP/902-007fe948 answered SIP/901-007f0e98 *Bridge about to start. Notice the correct channel names* [Jun 26 22:24:15] DEBUG[3668]: features.c:2483 ast_bridge_call: bridge answer set, chan answer set -- Packet2Packet bridging SIP/901-007f0e98 and SIP/902-007fe948 *AMI Redirect received* [Jun 26 22:24:19] DEBUG[11779]: manager.c:3007 process_message: Manager received command 'Redirect' [Jun 26 22:24:19] WARNING[11779]: channel.c:961 ast_channel_alloc_withId_withVaList: Sending Newchannel event with ActionID: (null) [Jun 26 22:24:19] DEBUG[11779]: channel.c:3980 ast_channel_masquerade: Planning to masquerade channel SIP/902-007fe948 into the structure of AsyncGoto/SIP/902-007fe948 [Jun 26 22:24:19] DEBUG[11779]: channel.c:3992 ast_channel_masquerade: Done planning to masquerade channel SIP/902-007fe948 into the structure of AsyncGoto/SIP/902-007fe948 [Jun 26 22:24:19] DEBUG[11779]: channel.c:4098 ast_do_masquerade: Actually Masquerading SIP/902-007fe948(6) into the structure of AsyncGoto/SIP/902-007fe948(6) [Jun 26 22:24:19] DEBUG[11779]: channel.c:4111 ast_do_masquerade: Got clone lock for masquerade on 'SIP/902-007fe948' at 0x805350 [Jun 26 22:24:19] DEBUG[11779]: channel.c:4292 ast_do_masquerade: Putting channel SIP/902-007fe948 in 8/8 formats [Jun 26 22:24:19] DEBUG[11779]: chan_sip.c:5512 sip_fixup: SIP Fixup: New owner for dialogue 0a0362e626aa6b5a0b3f3b3862f64...@10.10.1.213: SIP/902-007fe948 (Old parent: AsyncGoto/SIP/902-007fe948ZOMBIE) [Jun 26 22:24:19] DEBUG[11779]: channel.c:4338 ast_do_masquerade: Released clone lock on 'AsyncGoto/SIP/902-007fe948ZOMBIE' [Jun 26 22:24:19] DEBUG[11779]: channel.c:4347 ast_do_masquerade: Done Masquerading SIP/902-007fe948 (6) [Jun 26 22:24:19] DEBUG[11779]: channel.c:1576 ast_softhangup_nolock: Soft-Hanging up channel 'SIP/901-007f0e98' [Jun 26 22:24:19] DEBUG[3668]: rtp.c:4178 bridge_p2p_loop: p2p-rtp-bridge: Ooh, got a hangup *Returned from Bridge. Notice the incorrect channel name for the second channel* [Jun 26 22:24:19] DEBUG[3668]: channel.c:4921 ast_channel_bridge: Returning from native bridge, channels: SIP/901-007f0e98, AsyncGoto/SIP/902-007fe948ZOMBIE [Jun 26 22:24:19] DEBUG[3668]: channel.c:1675 ast_hangup: Hanging up zombie 'AsyncGoto/SIP/902-007fe948ZOMBIE' [Jun 26 22:24:19] DEBUG[3668]: rtp.c:2055 ast_rtp_early_bridge: Channel 'unspecified' has no RTP, not doing anything [Jun 26 22:24:19] DEBUG[3668]: app_dial.c:2032 dial_exec_full: Exiting with DIALSTATUS=ANSWER. [Jun 26 22:24:19] DEBUG[3668]: pbx.c:3779 __ast_pbx_run: Spawn extension (from-manager-core,calllegwait,1) exited non-zero on 'SIP/901-007f0e98' == Spawn extension (from-manager-core, calllegwait, 1) exited non-zero on 'SIP/901-007f0e98' [Jun 26 22:24:19] DEBUG[3668]: pbx.c:3179 pbx_extension_helper: Launching 'Wait' -- Executing [calllegw...@from-manager-core:1] Wait(SIP/901-007f0e98, 3600) in new stack [Jun 26 22:24:19] DEBUG[3670]: pbx.c:3179 pbx_extension_helper: Launching 'Wait' -- Executing [calllegw...@from-manager-core:1] Wait(SIP/902-007fe948, 3600) in new stack -- Regards, Prince Singh W:
Re: [asterisk-users] Echo and static on PRI with errors
Hi, I'm aware you didn't get a response. But please only post once, or at least leave a day or two. Is the PRI a known-good? i.e. tested with other stuff and error free? Steve On 30 Jun 2009, at 11:45, Tom O'Connor wrote: Hi there, I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. A little background: Server: HP DL145 G3 Dual Opteron 246 with 2GB ram and a brand new OpenVox D110P http://www.voipon.co.uk/openvox-d110p-pci-isdn-pri-card-p-658.html Dual 250GB SATA disks in Software Raid 1 Running Ubuntu 9.04 Jaunty, but I had the same problems on Intrepid and Hardy. Versions: Asterisk: 1.4.2, 1.6.0.10, 1.6.2beta4 libpri: 1.4.10 dahdi: 2.2.0-current Asterisk works fine for SIP calls, as long as they don't touch the outside world via the PRI card. This pastebin contains the console log from asterisk - vcg http://pastebin.com/f780c591e There are lots of chan_dahdi errors. Occasionally, it claims to run out of channels and terminates the active calls. This is the contents of /etc/dahdi/system.conf http://pastebin.com/f1f654235 This is the contents of /etc/asterisk/chan_dahdi.conf http://pastebin.com/f7ef35e72 This is /proc/interrupts http://pastebin.com/f61cd8398 This is lsmod http://pastebin.com/m56105bf5 I've tried stuff like binding the processor affinity of the modules to one or the other processor, I've tried changing the slot the card is in. I asked similar questions on #asterisk, and tried their suggestions. Nothing seems to work. Any help would be graciously recieved. I'm pretty much all out of ideas. Tom -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and static on PRI with errors.
On Tue, Jun 30, 2009 at 5:02 AM, Tom O'Connor t...@twinhelix.org wrote: Hi there, I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. A little background: Server: HP DL145 G3 Dual Opteron 246 with 2GB ram and a brand new OpenVox D110P http://www.voipon.co.uk/openvox-d110p-pci-isdn-pri-card-p-658.html Dual 250GB SATA disks in Software Raid 1 Running Ubuntu 9.04 Jaunty, but I had the same problems on Intrepid and Hardy. Versions: Asterisk: 1.4.2, 1.6.0.10, 1.6.2beta4 libpri: 1.4.10 dahdi: 2.2.0-current Asterisk works fine for SIP calls, as long as they don't touch the outside world via the PRI card. This pastebin contains the console log from asterisk -vcg http://pastebin.com/f780c591e There are lots of chan_dahdi errors. Occasionally, it claims to run out of channels and terminates the active calls. This is the contents of /etc/dahdi/system.conf http://pastebin.com/f1f654235 This is the contents of /etc/asterisk/chan_dahdi.conf http://pastebin.com/f7ef35e72 This is /proc/interrupts http://pastebin.com/f61cd8398 This is lsmod http://pastebin.com/m56105bf5 I've tried stuff like binding the processor affinity of the modules to one or the other processor, I've tried changing the slot the card is in. I asked similar questions on #asterisk, and tried their suggestions. Nothing seems to work. Any help would be graciously recieved. I'm pretty much all out of ideas. Tom -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org I would roll back to the latest version of 1.4 or even 1.2 with the latest version of good old Zaptel. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Opensips+asterisk problem
Hi Ram,
Does your OpenSIPS get any SIP reply from Asterisk? or the INVITE is
simply discarded by Asterisk?
Regards,
Bogdan
Hi all
After a long iam back to forum
back to my own topic and several readings done on this forum
how people doing same kind of setup what iam trying to achive
so here i have done some good developements
for testing iam doing all in one Server
Step1 :
Installed in Fresh BOX with Debian
Asterisk and A2B working Fine
Step2 : registered with SIP account iam able to make calls successfully
Step3 :
installed Opensips
Made Subscribers to view from A2b Database
Step4 : changed Asterisk port from 5060 to 5062
Step5 : Opensip config made changes to register users with Opensips
and when they dial 001X call send to Asterisk box
route[3]{
if (uri =~ sip:001[0...@*){
log(1, Forwarding to Asterisk \n);
rewritehostport(A2b-asterisk-IP:5062);
route(1);
exit;
}
Works Fine, No problems as of now
But to go in advance, i want to use Number of * boxes to achive more Load
Step5 : added Dispatcher Module in the Opensips
loadmodule dispatcher.so
.
.
.
modparam(dispatcher,list_file,/usr/local/etc/opensips/dispatcher.cfg)
.
.
.
.
changed route to use dispatcher
route[3]{
if (uri =~ sip:001[0...@*){
log(1, Forwarding to Asterisk \n);
ds_select_dst(2,4);
forward();
route(1);
exit;
}
My dispatcher Config Looks like below
dispatcher.cfg
2 sip:a2b-asterisk-ip:5062
2 sip:a2b-asterisk-ip2:5062
I have restarted Opensips
when i dial 0017XX number the call send Opensips to Asterisk
Jun 30 01:12:28 opensips[25868]: Forwarding to Asterisk
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: set [2]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: alg hash [1]
Jun 30 01:12:28 opensips[25868]: DBG:dispatcher:ds_select_dst: selected
[4-2/1] sip:a2b-asterisk-ip:5062
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:mk_proxy: doing DNS
lookup...
Jun 30 01:12:28 freeswitch opensips[25868]: DBG:core:forward_request:
sending:#012INVITE sip:0017x...@opensips-ip:5060
SIP/2.0#015#012Record-Route: sip:opensips-ip;lr=on#015#012Via: SIP/2.0/UDP
opensips-ip;branch=z9hG4bK28178282572929210914#015#012Via: SIP/2.0/UDP
ip-phone-ip:5060;received=ip-phone-ip;branch=z9hG4bK28178282572929210914;rport=5060#015#012From:
4720779942 sip:4720779...@opensips-ip:5060;tag=1966722825#015#012To:
0017325824631 sip:0017...@opensips-ip:5060#015#012Call-ID:
32167199575863-11502744529...@ip-phoneip#015#012cseq: 2
INVITE#015#012Contact:
sip:4720779...@ipphone-ip:5060#015#012Proxy-Authorization:
Digest username=4720779942, realm=asterisk, nonce=79ee65ba,
uri=sip:0017xxx...@opensips-ip:5060,
response=3e182f165a5663d0b145d6b55d34e94b,
algorithm=MD5#015#012Max-Forwards: 69#015#012Supported:
replaces#015#012User-Agent: Voip Phone 1.0#015#012Allow: INVITE, ACK,
OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE, PRACK,
UPDATE#015#012Content-Type: application/sdp#015#012Content-Length:
319#015#012#015#012v=0#015#012o=4720779942 17025328 32005127 IN IP4
202.63.111.2#015#012s=A conversation#015#012c=IN IP4 ip-phone-ip#015#012t=0
0#015#012m=audio 10028 RTP/AVP 18 4 8 0 9 101#015#012a=rtpmap:18
G729/8000#015#012a=rtpmap:4 G723/8000#015#012a=rtpmap:8
PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:9
G722/16000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101
0-15#015#012a=sendrecv#015#012.
opensips[25868]: DBG:core:forward_request: orig. len=1087, new_len=1220,
proto=1
when i ngrep
U 2009/06/30 01:59:20.770599 ipphone:5060 - asterisk-a2b-ip:5060
INVITE sip:0017x...@asterisk-a2b-ip:5060 SIP/2.0.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport.
From: 4720779942 sip:4720779...@asterisk-a2b-ip:5060;tag=3037030266.
To: 0017 sip:0017x...@asterisk-a2b-ip:5060.
Call-ID: 14399316162240-7371067914...@ipphone.
CSeq: 2 INVITE.
Contact: sip:4720779...@ipphone:5060.
Proxy-Authorization: Digest username=4720779942, realm=asterisk,
nonce=07ba8624, uri=sip:0017x...@asterisk-a2b-ip:5060,
response=5dbe9b2937d0bc3f6e8d25052fff0b6a, algorithm=MD5.
Max-Forwards: 70.
Supported: replaces.
User-Agent: Voip Phone 1.0.
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, SUBSCRIBE,
PRACK, UPDATE.
Content-Type: application/sdp.
Content-Length: 319.
.
v=0.
o=4720779942 69102627 18481147 IN IP4 ipphone.
s=A conversation.
c=IN IP4 ipphone.
t=0 0.
m=audio 10034 RTP/AVP 18 4 8 0 9 101.
a=rtpmap:18 G729/8000.
a=rtpmap:4 G723/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:9 G722/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.
U 2009/06/30 01:59:20.774528 asterisk-a2b-ip:5060 - ipphone:5060
SIP/2.0 100 Giving a try.
Via: SIP/2.0/UDP ipphone:5060;branch=z9hG4bK2932733762726732719;rport=5060.
From: 4720779942 sip:4720779...@asterisk-a2b-ip:5060;tag=3037030266.
To: 0017 sip:0017x...@asterisk-a2b-ip:5060.
Call-ID: 14399316162240-7371067914...@ipphone.
CSeq: 2 INVITE.
Server: OpenSIPS (1.5.1-notls
Re: [asterisk-users] Echo and static on PRI with errors.
On Tue, Jun 30, 2009 at 10:02:29AM +0100, Tom O'Connor wrote: Asterisk works fine for SIP calls, as long as they don't touch the outside world via the PRI card. This pastebin contains the console log from asterisk -vcg http://pastebin.com/f780c591e There are lots of chan_dahdi errors. Occasionally, it claims to run out of channels and terminates the active calls. This is the contents of /etc/dahdi/system.conf http://pastebin.com/f1f654235 Why have you disabled the echo canceller? This is the contents of /etc/asterisk/chan_dahdi.conf http://pastebin.com/f7ef35e72 This is /proc/interrupts http://pastebin.com/f61cd8398 This is lsmod http://pastebin.com/m56105bf5 dahdi_echocan_mg2 is loaded . However, what do you see on: dahdi show channel 1 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and static on PRI with errors.
On Tue, Jun 30, 2009 at 1:31 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Tue, Jun 30, 2009 at 10:02:29AM +0100, Tom O'Connor wrote: Asterisk works fine for SIP calls, as long as they don't touch the outside world via the PRI card. This pastebin contains the console log from asterisk -vcg http://pastebin.com/f780c591e There are lots of chan_dahdi errors. Occasionally, it claims to run out of channels and terminates the active calls. This is the contents of /etc/dahdi/system.conf http://pastebin.com/f1f654235 Why have you disabled the echo canceller? This is the contents of /etc/asterisk/chan_dahdi.conf http://pastebin.com/f7ef35e72 This is /proc/interrupts http://pastebin.com/f61cd8398 This is lsmod http://pastebin.com/m56105bf5 dahdi_echocan_mg2 is loaded . However, what do you see on: dahdi show channel 1 Oops. sorry about posting twice. The first one appeared to bounce. Anyway.. Either having the echo canceller enabled or disabled makes no difference. Steve, I was trying to avouid using 1.2, we're using 1.0 something at the moment, works fine, but new office forces a new server. Will 1.4 or 1.6 work with good old zaptel? Also, yes, the PRI works absolutely fine with the old asterisk server, and the settings are identical in the zaptel / system.conf files -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk module trouble
Hello, i have just installed asterisk 1.6.0.10 on debian 5.0 like: ./configure;make menuselect; make;make install There are no erorrs, but folder /usr/lib/asterisk/modules is empty. What am i doing wrong? Where are modules? p.s. Doing the same on Slackware, i ve got all selected modules at /usr/lib/asterisk/modules. -- Best regards, Maksim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and static on PRI with errors
On Tue, Jun 30, 2009 at 6:45 AM, Tom O'Connortom.bio...@gmail.com wrote: I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. In my experience, static and echo can be related to bad cables or bad physical telco wiring. This would explain why the problem affected every asterisk version you tried. Do you have any other gear you can terminate the PRI into for a comparison test? I would suspect a bad PRI. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk module trouble
On Tue, Jun 30, 2009 at 04:57:29PM +0400, M C wrote: Hello, i have just installed asterisk 1.6.0.10 on debian 5.0 like: ./configure;make menuselect; make;make install There are no erorrs, but folder /usr/lib/asterisk/modules is empty. What am i doing wrong? Where are modules? ls -l /usr/sbin/asterisk Any change you enabled module embedding? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.6 WaitForSilence Problem
I've set up an outbound .call system for customer callbacks and the
like. Calls are going out over analog lines and I'm trying to use the
WaitForSilence routine to make sure the phone has stopped ringing before
starting message playback. The problem is that if I set the first
argument of WaitForSilence to anything other than 1, WaitForSilence
never exits.
Some general info on my setup:
more /proc/version:
Linux version 2.6.16.60-0.34-smp (ge...@buildhost) (gcc version 4.1.2
20070115 (SUSE Linux)) #1 SMP Fri Jan 16 14:59:01 UTC 2009
Asterisk Version:
Connected to Asterisk 1.6.1.0 currently running on ivueivrtest (pid =
1639)
dahdi version:
2.2.0-rc4
/etc/dahdi/system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Thu May 21 11:50:14 2009
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: WCTDM/0 Wildcard TDM2400P Board 1 (MASTER)
fxsks=1
echocanceller=mg2,1
fxsks=2
echocanceller=mg2,2
fxsks=3
echocanceller=mg2,3
fxsks=4
echocanceller=mg2,4
# Global data
loadzone = us
defaultzone = us
/etc/asterisk/dahdi-channels.conf
; Autogenerated by /usr/sbin/dahdi_genconf on Thu May 21 11:50:15 2009
; If you edit this file and execute /usr/sbin/dahdi_genconf again,
; your manual changes will be LOST.
; Dahdi Channels Configurations (chan_dahdi.conf)
;
; This is not intended to be a complete chan_dahdi.conf. Rather, it is
intended
; to be #include-d by /etc/chan_dahdi.conf that will include the global
settings
;
; Span 1: WCTDM/0 Wildcard TDM2400P Board 1 (MASTER)
;;; line=1 WCTDM/0/0
threewaycalling=yes
callwaiting=yes
transfer=yes
callprogress=no
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 1
callerid=
group=
context=default
;;; line=2 WCTDM/0/1
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 2
callerid=
group=
context=default
;;; line=3 WCTDM/0/2
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 3
callerid=
group=
context=default
;;; line=4 WCTDM/0/3
signalling=fxs_ks
callerid=asreceived
group=0
context=from-pstn
channel = 4
callerid=
group=
context=default
/etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
IAXINFO=guest ; IAXtel
username/password
TRUNK=DAHDI/G0; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[outdial]
exten = s,1,Set(TIMEOUT(response)=3)
exten = s,n,WaitForSilence(4000,1,16)
exten = s,n,Agi(agi://localhost/Outdial.agi)
exten = s,n,Hangup()
exten = failed,1,Verbose(Outdial failed)
exten = failed,n,Verbose(Reason= ${REASON})
exten = failed,n,Hangup()
Thanks in advance for any help you can provide,
Deric Page
deric.p...@nisc.coop
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Re: [asterisk-users] Echo and static on PRI with errors
On Tue, Jun 30, 2009 at 2:16 PM, David Backeberg dbackeb...@gmail.comwrote: On Tue, Jun 30, 2009 at 6:45 AM, Tom O'Connortom.bio...@gmail.com wrote: I'm having some fairly serious asterisk problems, which seem to be spread quite liberally across all asterisk versions, I've tried 1.4.2, 1.6.0.10, 1.6.2beta4 and still had exactly the same problem with static and echo on the line when using the PRI interface. In my experience, static and echo can be related to bad cables or bad physical telco wiring. This would explain why the problem affected every asterisk version you tried. Do you have any other gear you can terminate the PRI into for a comparison test? I would suspect a bad PRI. I'd love to accept that as the cause, but we've got an old asterisk box, sapphire*CLI show version Asterisk 1.0-RC1 built by r...@nyx on a i686 running Linux Which connects and works fine, no PRI errors. The problem is, we want to do stuff that isn't supported by that old version, MoH doesn't work properly. For the SIP part, the 1.6.2beta is perfect. I'm currently pointing fingers at either the hardware (someone on #asterisk said it could be a cruddy chipset, but it's an HP Server.. so should be kosher.. ), I might try a stock kernel, instead of an ubuntu one.. but there's a bit of FUD involved there. I might try a totally different server also.. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk module trouble
2009/6/30 Tzafrir Cohen tzafrir.co...@xorcom.com On Tue, Jun 30, 2009 at 04:57:29PM +0400, M C wrote: Hello, i have just installed asterisk 1.6.0.10 on debian 5.0 like: ./configure;make menuselect; make;make install There are no erorrs, but folder /usr/lib/asterisk/modules is empty. What am i doing wrong? Where are modules? ls -l /usr/sbin/asterisk Any change you enabled module embedding? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Debian-50-lenny-32-minimal:/usr/lib/asterisk/modules# ls -l /usr/sbin/asterisk -rwxr-xr-x 1 root root 39M 2009-06-30 14:29 /usr/sbin/asterisk Yes, i ve embeded all modules in menuselect. Also, i`ve installed Asterisk before on Slackware and there was not such error. -- Best regards, Maksim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk module trouble
There are no erorrs, but folder /usr/lib/asterisk/modules is empty. What am i doing wrong? Where are modules? Yes, i ve embeded all modules in menuselect Uh.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with DTMF detection in ast_app_getdata (*1.2)
Hi, I am using a basic VOIP phone (find here: http://www.tootoo.com/d-rp20207560-VoIP_phone/) on an Asterisk 1.2.26-BRIstuffed-0.3.0-PRE-1y-q version of asterisk. I am running a C-based prepaid application based on MySQL that accepts dtmf events from the phone to authenticate. When asterisk is configured with DTMF mode rfc 2833 or auto (or any of the others in fact) the ast_app_getdata method sometimes just times out and doesn't get the DTMF events from the phone (the phone also has support for rfc 2833 or inband). However when using auto or 2833, my software Twinkle client will ALWAYS get its DTMF events detected without fail (so far). Has anyone come across this situation before, I have exhausted the options between Asterisk and my SIP phone DTMF settings. BTW I also use rlaxdtmf=yes in sip conf. Thanks Mos ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and static on PRI with errors
On Tuesday 30 June 2009 08:24:29 Tom O'Connor wrote: I'm currently pointing fingers at either the hardware (someone on #asterisk said it could be a cruddy chipset, but it's an HP Server.. so should be kosher.. ), I Is it an HP server from the HP server line, or is it an HP server from the old Compaq line? Don't assume that because of the HP name, it's actually reliable with 3rd party hardware. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Setting CDR(userfield) from Macro called from feature doesn't work with cdr_mysql
cdr_mysql doesn't set the userfield when it's set inside a macro
called from a feature (1.4.25, addons 1.4.8).
I have a feature code:
autorecord = *1,self,Macro,apprecord
The apprecord macro looks like:
[macro-apprecord]
exten = s,1,Playback(beep)
exten =
s,n,Set(RECORDFILE=/var/spool/asterisk/autorecord/${STRFTIME(${EPOCH},,%Y/%m/%d/%H%M%S)}-${UNIQUEID}-^-${CALLERID(num)})
exten = s,n,Set(CDR(userfield)=${RECORDFILE})
exten = s,n,MixMonitor(${RECORDFILE}.wav)
exten = s,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = s,n,NoOp(CDR(userfield) = ${CDR(userfield)})
exten = s,n,MacroExit
The NoOp shows the userfield is set correctly but the userfield is blank
in my MySQL cdr database. I set CDR(userfield) elsewhere in the dialplan
and this works so it seems to be related to being set within a macro.
Any idea what I'm doing wrong?
--
Regards,
Russell
| Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] IAX2 help needed...
I run a phone in a remote office using the IAX2 protocol. It mostly works fine; except that every 5 mins it loses connection with Asterisk, before reconnecting 30 seconds later; rinse repeat. Using the IAX2 debugging, I'm seeing this a lot: Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00018ms SCall: 04050 DCall: 0 [**.**.***.***:4673] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00018ms SCall: 16174 DCall: 04050 [**.**.***.***:4673] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00018ms SCall: 16174 DCall: 04050 [**.**.***.***:4673] RR_JITTER : 0 Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00018ms SCall: 04050 DCall: 16174 [**.**.***.***:4673] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00018ms SCall: 16174 DCall: 04050 [**.**.***.***:4673] RR_JITTER : 0 Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 04050 DCall: 16174 [**.**.***.***:4673] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 16175 DCall: 0 [**.**.***.***:4673] USERNAME: 5111 REFRESH : 60 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGACK Timestamp: 00019ms SCall: 08339 DCall: 16175 [**.**.***.***:4673] USERNAME: 5111 DATE TIME : 2009-06-30 15:27:40 REFRESH : 60 APPARENT ADDRES : IPV4 **.**.***.***:4673 CALLING NUMBER : 5111 CALLING NAME: Ade Vickers (home) Note in particular: Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 04050 DCall: 16174 [**.**.***.***:4673] Whenever this happens, the phone loses connection until a REGACK is received. This started happening when I upgraded Asterisk to v 1.4.22 (from an earlier v1.4.x), on a new machine. Any ideas what I need to do to fix the issue? Phone is a Quartel 710E, in case that's of any use, and it worked fine with my previous Asterisk setup. Cheers, Ade. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial Chan_local Usage
Elliot Murdock murdo...@gmail.com writes: I needed to answer the local call for any sound to pass through: [default] exten = s,1,dial(local/2...@dialplan/n) [dailplan] exten = 220,1,answer() exten = 220,2,saydigits(123) exten = 220,3,dial(SIP/120||m) From my understanding, the answer command only answers the local call, but the final dial at priority 3 will remain unanswered. I guess you could put it that way, but notice that the original caller will start paying the moment you Answer(). Playing sounds before Answer() is called early media. It is unfortunately not universally supported -- possibly because it is so easily abused. Just imagine having two sets of phones, both transmitting early media. That would mean free calls. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and static on PRI with errors
On Tue, Jun 30, 2009 at 3:12 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com wrote: On Tuesday 30 June 2009 08:24:29 Tom O'Connor wrote: I'm currently pointing fingers at either the hardware (someone on #asterisk said it could be a cruddy chipset, but it's an HP Server.. so should be kosher.. ), I Is it an HP server from the HP server line, or is it an HP server from the old Compaq line? Don't assume that because of the HP name, it's actually reliable with 3rd party hardware. It's a HP DL145 G2. more than that, i can't say. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting CDR(userfield) from Macro called from feature doesn't work with cdr_mysql
Check this issue, seems related
https://issues.asterisk.org/view.php?id=14662
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russell Brown
Sent: martes, 30 de junio de 2009 11:33 a.m.
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Setting CDR(userfield) from Macro called from
feature doesn't work with cdr_mysql
cdr_mysql doesn't set the userfield when it's set inside a macro
called from a feature (1.4.25, addons 1.4.8).
I have a feature code:
autorecord = *1,self,Macro,apprecord
The apprecord macro looks like:
[macro-apprecord]
exten = s,1,Playback(beep)
exten =
s,n,Set(RECORDFILE=/var/spool/asterisk/autorecord/${STRFTIME(${EPOCH},,%Y/%m
/%d/%H%M%S)}-${UNIQUEID}-^-${CALLERID(num)})
exten = s,n,Set(CDR(userfield)=${RECORDFILE})
exten = s,n,MixMonitor(${RECORDFILE}.wav)
exten = s,n,Set(AUDIOHOOK_INHERIT(MixMonitor)=yes)
exten = s,n,NoOp(CDR(userfield) = ${CDR(userfield)})
exten = s,n,MacroExit
The NoOp shows the userfield is set correctly but the userfield is blank
in my MySQL cdr database. I set CDR(userfield) elsewhere in the dialplan
and this works so it seems to be related to being set within a macro.
Any idea what I'm doing wrong?
--
Regards,
Russell
| Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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Re: [asterisk-users] Calling non-extension numbers issue
Hi, Am Montag, den 29.06.2009, 10:35 -0400 schrieb Kayton Sapale: That's the strange thing. Nothing shows when monitoring the service in debug. On the phone, however, I do see a connection time-out error. I guess this might indicate that the device is attempting to connect to the service in a way different from when just dialing an extension? If You don't see anything on the command line of *, there might be an issue with Your phone settings. I don't know anything about the nokias, but I *think* it might be possible, that the phone connects to anything other than Your * box in case of the outbond number. AFAIK the * sends a 404-Error back on an non existing extension. In this case the phone would not show up a connection time-out. So I would check the settings on the phone. Or maybe You could do a network trace with tcpdump or ngrep to double check, that the phone really tries to connect to *. HTH, Karsten ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo and static on PRI with errors
Tom O'Connor wrote: On Tue, Jun 30, 2009 at 3:12 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com mailto:tilgh...@mail.jeffandtilghman.com wrote: On Tuesday 30 June 2009 08:24:29 Tom O'Connor wrote: I'm currently pointing fingers at either the hardware (someone on #asterisk said it could be a cruddy chipset, but it's an HP Server.. so should be kosher.. ), I Is it an HP server from the HP server line, or is it an HP server from the old Compaq line? Don't assume that because of the HP name, it's actually reliable with 3rd party hardware. It's a HP DL145 G2. more than that, i can't say. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org mailto:t...@twinhelix.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The card is TE110P compatible and as such probably suffers from the same interrupt sharing problem. The ...HDLC Bad FCS.. messages tend to be related to interrupt sharing. What does lspci -vb show? Anything sharing interrupts with the card? Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question regarding SIP 183 Session Progress handling in Asterisk
Floimair Florian schrieb: I am having trouble with a project concerning the 183 Session Progress SIP messages. Asterisk seems to only accept these when there is also a Session Description (SDP) included in the message. I also verified this by looking at the code. Which version of Asterisk? However for a project we are working with a trunk to a third party system (Alcatel) and they are insisting that this behavior is non-compliant with RFC3261 (SIP). So can someone please tell me the reason, why Asterisk does not support 183 messages without SDP as this would really help me finding arguments in this situation. So far Alcatel just tells us that this is not SIP-compliant and that we have to change things on the Asterisk side, but I'm not quite sure that this is really the case and having arguments could help me clarifying this situation. What they tell you might actually be correct. Not sure. Sip: f.floim...@commend.com file:///T:\KAT\Signaturen\=%22sip:f.floim...@commend.com%22 Something went wrong here. JFYI. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Reception of vocal SMSs to landlines.
Hi all, we face a problem with SMS reception sended to _landlines_, at least in France. Normally operators -tested with France Telecom and SFR- are sending voice SMSs from a particular CID number, so no problem. But today we discover that -at least SFR- send from time to time voice SMSs with original callerID which means that the call is terminated like a normal call and not recognized as voice SMS. Problem is that we Answer() the call or we forward it or we send it to voicemail, each user having his own setup. In such cases, the SMS will _never_ be delivered as: . after we Answer() the call, the operator send immediately the audio SMS. At this time we are just parsing dialplan to see what to do with the call. If the call is to forward, operator hangup -end of vocal SMS- before the called party could take the call. . if sending to voicemail, with or without Answer(), and as generally you send audio before recording (eg Our office is currently close, blabla ...), during this time the SMS is already readed! You just catch the last audio from operator message if introduction is not to long. . if you Answer() and don't take the call (busy or second line or ...) same that for VM: you catch nothing or only the end of message. Is there a way to detect voice audio before to -at least- send directly to a voicemail without announcement? Or is another solution existing? Thanks for any hint -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reception of vocal SMSs to landlines.
This is not a clean or efficient solution, but you could use an AGI or .call file to sent the SMS as a separate call. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Tuesday, June 30, 2009 12:43 PM To: Asterisk-Users Subject: [asterisk-users] Reception of vocal SMSs to landlines. Hi all, we face a problem with SMS reception sended to _landlines_, at least in France. Normally operators -tested with France Telecom and SFR- are sending voice SMSs from a particular CID number, so no problem. But today we discover that -at least SFR- send from time to time voice SMSs with original callerID which means that the call is terminated like a normal call and not recognized as voice SMS. Problem is that we Answer() the call or we forward it or we send it to voicemail, each user having his own setup. In such cases, the SMS will _never_ be delivered as: . after we Answer() the call, the operator send immediately the audio SMS. At this time we are just parsing dialplan to see what to do with the call. If the call is to forward, operator hangup -end of vocal SMS- before the called party could take the call. . if sending to voicemail, with or without Answer(), and as generally you send audio before recording (eg Our office is currently close, blabla ...), during this time the SMS is already readed! You just catch the last audio from operator message if introduction is not to long. . if you Answer() and don't take the call (busy or second line or ...) same that for VM: you catch nothing or only the end of message. Is there a way to detect voice audio before to -at least- send directly to a voicemail without announcement? Or is another solution existing? Thanks for any hint -- Daniel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Intercepting a Call while ringing a device
Hello! I am looking for a way to dynamically redirect a call while it is ringing to another device. Basically, if a person is far away from his desk, he should have the option to use another phone and pick up the call. Thanks for any suggestions, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intercepting a Call while ringing a device
If it is configured and working correctly, *8 picks up the ringing line from any eligible phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock Sent: Tuesday, June 30, 2009 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Intercepting a Call while ringing a device Hello! I am looking for a way to dynamically redirect a call while it is ringing to another device. Basically, if a person is far away from his desk, he should have the option to use another phone and pick up the call. Thanks for any suggestions, Elliot ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Intercepting a Call while ringing a device
Elliot Murdock schrieb: I am looking for a way to dynamically redirect a call while it is ringing to another device. Basically, if a person is far away from his desk, he should have the option to use another phone and pick up the call. Pickup() application? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension status as XML for an Aastra 57i
I'm in the process of converting our current hybrid key system to Asterisk and Aastra 57i phones. One of the features that seems to be a show stopper for almost everyone in the office is the inability to see who is on the phone. Can someone point in the right direction to setup an XML app on the phone so they can press a soft-button and get a list of extensions and their statuses? I know I can use BLF and the line 2-4 buttons; but there are a lot more then 3 other people working here and I'm planning on using those of parking lots. Any help will be greatly appreciated as I'm an Asterisk noob learning as fast as I can. Thanks in advance, Jeremy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension status as XML for an Aastra 57i
On Tue, Jun 30, 2009 at 4:17 PM, Jeremy Winder jwin...@logicalsi.comwrote: I'm in the process of converting our current hybrid key system to Asterisk and Aastra 57i phones. One of the features that seems to be a show stopper for almost everyone in the office is the inability to see who is on the phone. Can someone point in the right direction to setup an XML app on the phone so they can press a soft-button and get a list of extensions and their statuses? I know I can use BLF and the line 2-4 buttons; but there are a lot more then 3 other people working here and I'm planning on using those of parking lots. Any help will be greatly appreciated as I'm an Asterisk noob learning as fast as I can. Thanks in advance, Jeremy Side cars I guess. You could give them all a shortcut to FOP (Flash Operator Panel) I hear snom makes a good sidecar but I have zero experience with them. Another thought is to integrate with Jabber so you can see who it on the phone or even away, or whatever status. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension status as XML for an Aastra 57i
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Jeremy Winder wrote: I'm in the process of converting our current hybrid key system to Asterisk and Aastra 57i phones. One of the features that seems to be a show stopper for almost everyone in the office is the inability to see who is on the phone. Can someone point in the right direction to setup an XML app on the phone so they can press a soft-button and get a list of extensions and their statuses? I know I can use BLF and the line 2-4 buttons; but there are a lot more then 3 other people working here and I'm planning on using those of parking lots. Any help will be greatly appreciated as I'm an Asterisk noob learning as fast as I can. If you'd like a more generalized approach you can install an Openfile server and use the Asterisk plugin. That'll give you an internal IM server which will show the status you seek. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKSnZnCFu3bIiwtTARAveUAKCACUaxvBLfYHtuhojOZW1o/aOVkACdH3EQ ceTOXQOXlENUTGWhevxIrXc= =rG0d -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe not prompting for PIN
Hello, all. I must be brain cramping badly on our Asterisk 1.6.1.1 installation. Our MeetMe macros are working fine except they do not prompt for a PIN. So I made a very simple conference room: exten = ,1,MeetMe(123456,cMaAsx,123456) Shouldn't this prompt the user who dials to enter a PIN before entering the conference room whether or not a PIN is defined in meetme.conf? I have tried it both ways and tried using the P flag. The user is never prompted. What am I missing? Thanks - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Adit 600 Configuration
Has anyone ever gotten an Adit 600 to work with Asterisk1.4 via MGCP.
Asterisk keeps giving me the following error in the LOGs:
[Jun 30 08:32:59] NOTICE[26785]: chan_mgcp.c:1726
find_subchannel_and_lock: Gateway 10.0.0.245' (and thus its endpoint
'*') does not exist
MGCP Config:
[AFSWestAdit600]
host = dynamic
context = default
canreinvite = no
threewaycalling = yes
cancallforward = yes
transfer = yes
callwaiting = yes
slowsequence = yes
line = aaln/3
line = aaln/2
line = aaln/1
Extensions.conf
[default]
exten = 3412,1,Dial(MGCP/aaln/1...@afswestadit600)
exten = 3413,1,Dial(MGCP/aaln/1...@afswestadit600)
exten = 3414,1,Dial(MGCP/aaln/1...@afswestadit600)
Adit Config
set verification off
set 6 autologout 0
-set 6 password view {password} is manual
-set 6 password config {password} is manual
-set 6 password admin {password} is manual
-set 6 enhanced security enable is manual
-set 6 password security {password} is manual
set 6 priority tos 0xFC 0xB8
delete 6 remote RemoteUnit
set 6:1 framing ipx ieee8023 disable
set 6:1 framing ipx ieee8022 disable
set 6:1 framing ipx snap disable
set 6:1 framing ipx ethii disable
set 6:1 ip address 10.0.0.245 255.255.255.0
set 6:1 gateway 10.0.0.1
set 6 dns domain local.local
set 6 dns name afswestadit600
set 6 dns server 1 10.0.0.135
set 6 dns resolver enable
set 6:1 up
add 6:1 static ip network 10.0.0.0 255.255.255.0 10.0.0.1 1
add 6 remote RemoteUnit
set 6 snmp name unknown
set 6 snmp contact unknown
set 6 snmp location unknown
set 6 RemoteUnit up
set 6 log last detail
set 6 mgcp callagent address 10.0.0.167
set 6 mgcp gatewayid 10.0.0.245
set 6 mgcp quarantine step discard
set 6 mgcp port 2727
set 6 mgcp up
set 6 mgcp rsipwildcard enable
set 6 mgcp tos 0x68
set 6 voip osi 500
set 6:1:1:1 log start both
set 6:1:1:1-48 echo tail 64
set 6:1:1:1-48 tos 0xB8
set 6:1:1:1-48 algorithm preference g711mu g729a
set 6:1:1:1-48 dtmfrelay enable
set 6:1:1:1-48 cpd osi
reset 6
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Re: [asterisk-users] Extension status as XML for an Aastra 57i
On Tue, Jun 30, 2009 at 3:17 PM, Jeremy Winderjwin...@logicalsi.com wrote: I'm in the process of converting our current hybrid key system to Asterisk and Aastra 57i phones. One of the features that seems to be a show stopper for almost everyone in the office is the inability to see who is on the phone. Can someone point in the right direction to setup an XML app on the phone so they can press a soft-button and get a list of extensions and their statuses? We have a few of those phones and the associated sidecards. Using a combination of hints and parking, you can get the status of the other phones to show up on the side car. I found all the information on voip-info.org for how to do it. Parking, hints the aastra configs and all. You can also do that with the softkeys on the top and bottom of the screen. -jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension status as XML for an Aastra 57i
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Barry L. Kline wrote: If you'd like a more generalized approach you can install an Openfile server and use the Asterisk plugin. That'll give you an internal IM server which will show the status you seek. Sorry, not 'openfile' but 'openfire'. http://www.igniterealtime.org/projects/openfire/index.jsp -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFKSntACFu3bIiwtTARAp4NAJ9/SdVaNXlc6nNq3LB8A5ss/X2q8wCcDNd0 Cq6E39xouEePRfGgqZULYzo= =kbf5 -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MeetMe not prompting for PIN
No That says: Join Conference 123456 The PIN for the Conference is 123456 What you need to do is setup the conference ID, guest PIN, admin PIN in the meetme.conf and then use exten = ,1,MeetMe(123456,cMaAsxp,) John A. Sullivan III wrote: Hello, all. I must be brain cramping badly on our Asterisk 1.6.1.1 installation. Our MeetMe macros are working fine except they do not prompt for a PIN. So I made a very simple conference room: exten = ,1,MeetMe(123456,cMaAsx,123456) Shouldn't this prompt the user who dials to enter a PIN before entering the conference room whether or not a PIN is defined in meetme.conf? I have tried it both ways and tried using the P flag. The user is never prompted. What am I missing? Thanks - John -- *Singer X.J. Wang* /System and Database Engineer/ The Pythian Group Office: (613) 565-8696 x298 Toll Free: (877) 798-4426 x298 Fax:(613) 565-8710 Email: w...@pythian.com MSN:pythianw...@hotmail.com Yahoo: pythianwang AIM:pythianwang ICQ:201253 Gadu-Gadu: 6817795 Tencent QQ: 858310404 begin:vcard fn:Singer Wang n:Wang;Singer org:The Pythian Group;Team 13 adr:116 Albert Street;;Suite 1000;Ottawa;Ontario;K1P 5G3;Canada email;internet:w...@pythian.com title:System and Database Administrator tel;work:(613) 565-8696 x298 tel;fax:(613) 565-8710 x-mozilla-html:TRUE url:http://www.pythian.com version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Extension status as XML for an Aastra 57i
On Tue, 2009-06-30 at 16:17 -0400, Jeremy Winder wrote: I'm in the process of converting our current hybrid key system to Asterisk and Aastra 57i phones. One of the features that seems to be a show stopper for almost everyone in the office is the inability to see who is on the phone. Can someone point in the right direction to setup an XML app on the phone so they can press a soft-button and get a list of extensions and their statuses? I know I can use BLF and the line 2-4 buttons; but there are a lot more then 3 other people working here and I'm planning on using those of parking lots. Any help will be greatly appreciated as I'm an Asterisk noob learning as fast as I can. The 57i phone has 6 soft buttons which can show the status of at least 16 phones (if you do not want to use the rest of the soft buttons which would give you another 16). If you really need to have more you should use the 536M or 560M console which can display up to 60 extensions. -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Authentication Issue Between Servers
I've got an issue where I am trying to route calls between Asterisk
Servers. I can route calls inbound to a server but seem to have an
authentication issue going out over the same sip account. It appears
that my server isn't sending the second invite after proxy
authentication request. I can't figure out why; any ideas would be
greatly appreciated. Thanks!
- Josh
Here is my sip.conf:
[general]
context = default
allowoverlap = no
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes
externip = 172.21.235.2
localnet = 172.21.235.2/255.255.0.0
dtmfmode = rfc2833
relaxdtmf = yes
disallow = all
allow = ulaw
allow = gsm
maxexpirey = 30
defaultexpirey = 180
relaxdtmf=yes
canreinvite = no
nat = 0
UserAgent = Asterisk
echocancel = yes
echocancelwhenbridge = yes
t38pt_udptl = no
[trunk]
type = friend
callwaiting = yes
caller id =
contact =
context = default
fullname =
group =
hasagent = no
hasdirectory = yes
hasiax = no
hasmanager = no
hassip = yes
host = 172.21.235.1
secret = [password]
threewaycalling = yes
registersip = yes
canreinvite = no
nat = no
dtmfmode = rfc2833
registeriax = no
disallow = all
allow = gsm
register=trunk:[passwo...@172.21.235.1
Here is the applicable portion of extensions.conf:
[default]
exten = _5XX,1,Dial(SIP/trunk/${EXTEN},,Tt)
Here is the SIP Debug output:
INVITE sip:5...@172.21.235.1 SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: Marci sip:3...@172.21.235.2;tag=as5951033c
To: sip:5...@172.21.235.1
Contact: sip:3...@172.21.235.2
Call-ID: 430c49156ce4a7500b1fa57807b5a...@172.21.235.2
CSeq: 102 INVITE
User-Agent: Asterisk
Max-Forwards: 70
Date: Tue, 30 Jun 2009 19:09:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 239
v=0
o=root 11411 11411 IN IP4 172.21.235.2
s=session
c=IN IP4 172.21.235.2
t=0 0
m=audio 11486 RTP/AVP 3 101
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
^@
^[[KWBPBXFG000304*CLI
--- SIP read from 172.21.235.1:5060 ---
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
172.21.235.2:5060;branch=z9hG4bK78d4e8d7;received=172.21.235.2;rport=5060
From: Marci sip:3...@172.21.235.2;tag=as5951033c
To: sip:5...@172.21.235.1;tag=as045cd609
Call-ID: 430c49156ce4a7500b1fa57807b5a...@172.21.235.2
CSeq: 102 INVITE
User-Agent: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=4c4374da
Content-Length: 0
-
^@
^[[KWBPBXFG000304*CLI
--- (11 headers 0 lines) ---
^@
^[[KWBPBXFG000304*CLI
Transmitting (NAT) to 172.21.235.1:5060:
ACK sip:5...@172.21.235.1 SIP/2.0
Via: SIP/2.0/UDP 172.21.235.2:5060;branch=z9hG4bK78d4e8d7;rport
From: Marci sip:3...@172.21.235.2;tag=as5951033c
To: sip:5...@172.21.235.1;tag=as045cd609
Contact: sip:3...@172.21.235.2
Call-ID: 430c49156ce4a7500b1fa57807b5a...@172.21.235.2
CSeq: 102 ACK
User-Agent: Asterisk
Max-Forwards: 70
Content-Length: 0
---
^@
^[[KWBPBXFG000304*CLI
[Jun 30 14:09:25] NOTICE[11434]: chan_sip.c:12253
handle_response_invite: ^...@failed to authenticate on INVITE to 'Marci
sip:3...@172.21.235.2;tag=as5951033c'
^@
^[[KWBPBXFG000304*CLI
Really destroying SIP dialog
'430c49156ce4a7500b1fa57807b5a...@172.21.235.2' Method: INVITE
^@
^[[KWBPBXFG000304*CLI
Really destroying SIP dialog
'0fe5f50f7674160d2ab3522f09060...@127.0.0.1' Method: REGISTER
^@
^[[KWBPBXFG000304*CLI
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[asterisk-users] Puzzling problem
Hi All, I have a problem with my Asterisk Server that the logs aren't giving me any clue to what's going on. The server is running 1.6.1.1 and connected to a Grandstream GXP2000 phone. At 3:58 minutes the call cuts off with no indication in the log. This is random and is only localized to that 1 phone. The other phone is a cordless connected through a Sipura Box with no problems. I've tried other versions of Asterisk after the problem started and it is continuing. Any help on where to look for clues is greatly appreciated. TIA, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using DIALSTATUS question
Thanks again Jim. I seem to be successful in using this method but now I get
the following after the call completes. It seems that asterisk doesn't know
what to do with the first channel. Would this indicate I am missing a
Hangup() somewhere?
Thx.
:
[Jun 30 18:31:30] WARNING[26484]: pbx.c:3907 __ast_pbx_run: Don't know what
to do with 'Local/dialnum...@mycompany_cdi_private-3232;1'
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Dickenson
Sent: Friday, June 26, 2009 1:09 PM
To: Asterisk User MailList
Subject: Re: [asterisk-users] Using DIALSTATUS question
I am using version 1.6.0.x and you can do core show application dial at
CLI to see info about the dial command.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
_
From: John Regal jre...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Fri, 26 Jun 2009 12:32:19 -0400
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Using DIALSTATUS question
Thanks so much for this method. I am going to give it a shot. I am not
familiar with that ghM part. I tried looking for information on it - Is
that some undocumented macro call feature or something?
Thanks again.
John
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
mailto:asterisk-users-boun...@lists.digium.com%5d On Behalf Of Jim
Dickenson
Sent: Wednesday, June 03, 2009 11:19 PM
To: Asterisk User MailList
Subject: Re: [asterisk-users] Using DIALSTATUS question
They way I do dialing is with this AMI packet:
Action: Originate
Channel: Local/dial_num...@cfmc_cdi_private
Exten: 1322
Context: default
Priority: 1
Variable: CfMC_ActionID=callE1321
Variable: CfMC_DialInfo=Dahdi/G1/8881231234
Variable: CfMC_RingTimeout=30
ActionID: callE1321
Async: true
And these extensions:
[macro-cfmc_dial_private]
exten = s,1,UserEvent(DidDial,ActionID:${ARG1} ${UNIQUEID} ${CHANNEL}
${ARG2})
[cfmc_cdi_private]
exten = dial_number,1,UserEvent(BeforeDial,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_DialInfo} ${CfMC_RingTimeout})
exten =
dial_number,n,Dial(${CfMC_DialInfo},${CfMC_RingTimeout},ghM(cfmc_dial_privat
e^${CfMC_ActionID}^${CfMC_DialInfo}))
; DIALSTATUS - CHANUNAVAIL CONGESTION NOANSWER BUSY ANSWER CANCEL DONTCALL
TORTURE INVALIDARGS
exten = dial_number,n,UserEvent(AfterDial,ActionID:${CfMC_ActionID}
${UNIQUEID} ${CHANNEL} ${CfMC_DialInfo} ${DIALSTATUS})
exten = dial_number,n,Hangup()
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: John Regal jre...@gmail.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Wed, 3 Jun 2009 14:38:09 -0400
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Subject: [asterisk-users] Using DIALSTATUS question
Hi all,
I am trying to make decisions in my dialplan based on {DIALSTATUS}. I am
creating calls using AMI (rawman with parameters via URL) with
action:Originate. I am using SIP and an outside voip provider for the calls.
If I define the number to call in the Channel parameter (e.g.
SIP/1555...@myvoipprovider, the call gets placed before entering the
context that I defined. I understand that the call only gets put into the
context if the call was answered. If the voip provider returns a busy code,
I cannot test for it in the dialplan since it never entered the context I
defined in the Originate command. Calls that are answered and therefore make
it into the dialplan show {DIALSTATUS} as null (when I echo it from the
context).
How can I programmatically place calls and evaluate dialstatus using SIP?
My sip.conf looks like this:
[general]
disallow=all
allow=ulaw
allow=g729
register = username:sec...@170.17.13.13
[myvoipprovider]
type=friend
secret=secret
username=username
host=sip.myvoipprovider.com
dtmfmode=rfc2833
context=outbound
qualify=yes
canreinvite=no
allow=ulaw
allow=g729
insecure=port,invite
Thanks.
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Re: [asterisk-users] Puzzling problem
I have had weird issues with that model, too. Have you tried reseting the phone to factory defaults and then reconfigure? The directions to reset can be found here: http://sipx-wiki.calivia.com/index.php/HowTo_configure_Grandstream_SIP_phone _with_sipX hope this helps -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese Sent: Tuesday, June 30, 2009 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Puzzling problem Hi All, I have a problem with my Asterisk Server that the logs aren't giving me any clue to what's going on. The server is running 1.6.1.1 and connected to a Grandstream GXP2000 phone. At 3:58 minutes the call cuts off with no indication in the log. This is random and is only localized to that 1 phone. The other phone is a cordless connected through a Sipura Box with no problems. I've tried other versions of Asterisk after the problem started and it is continuing. Any help on where to look for clues is greatly appreciated. TIA, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Puzzling problem
Try upgrading the firmware on it. They have all sorts of goofy bugs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese Sent: Tuesday, June 30, 2009 4:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Puzzling problem Hi All, I have a problem with my Asterisk Server that the logs aren't giving me any clue to what's going on. The server is running 1.6.1.1 and connected to a Grandstream GXP2000 phone. At 3:58 minutes the call cuts off with no indication in the log. This is random and is only localized to that 1 phone. The other phone is a cordless connected through a Sipura Box with no problems. I've tried other versions of Asterisk after the problem started and it is continuing. Any help on where to look for clues is greatly appreciated. TIA, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Puzzling problem
I did the upgrade to the phone. And the problem continued. Currently, as per the previous poster, I have reset the phone to the factory default and have started setup again. Peder wrote: Try upgrading the firmware on it. They have all sorts of goofy bugs. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Todd Reese Sent: Tuesday, June 30, 2009 4:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Puzzling problem Hi All, I have a problem with my Asterisk Server that the logs aren't giving me any clue to what's going on. The server is running 1.6.1.1 and connected to a Grandstream GXP2000 phone. At 3:58 minutes the call cuts off with no indication in the log. This is random and is only localized to that 1 phone. The other phone is a cordless connected through a Sipura Box with no problems. I've tried other versions of Asterisk after the problem started and it is continuing. Any help on where to look for clues is greatly appreciated. TIA, Todd Reese ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls dropping
Hi,
I am using Originate in testing and also using call files in testing. I also
needed to capture DIALSTATUS and update my CDRs accordingly. My original
attempts at using Originate (or call files) did not report {DIALSTATUS} if
the call could not be connected (e.g. bad phone number like 555-555-)
because, it seems, the call never entered the dialplan context defined in
Originate. So, my latest code uses Local/mycontext/n
So it appears, intermittently, the channel disappears (fails?). This is the
output of an instance of failure. I have also listed the code from my
context and the AMI call. A more readable version is attached. I am 1.6.1.1
- Thanks for looking...
-- Executing [dialnum...@dialthem_private:2]
Dial(Local/dialnum...@dialthem_private-c64a;2,
SIP/1211...@flowroute,30,ghM(dialthem_private^callJohnSmith^SIP/121
1...@flowroute)) in new stack
== Using SIP RTP CoS mark 5
-- Called 1211...@flowroute
-- SIP/flowroute-0821ffc8 is making progress passing it to
Local/dialnum...@dialthem_private-c64a;2
== Spawn extension (dialthem_private, dialnumber, 2) exited non-zero on
'Local/dialnum...@dialthem_private-c64a;2'
[Jun 30 19:52:01] ERROR[26664]: pbx.c:8637 device_state_cb: Received invalid
event that had no device IE
[Jun 30 19:52:01] ERROR[26664]: app_queue.c:810 device_state_cb: Received
invalid event that had no device IE
[dialthem_private]
exten = dialnumber,1,UserEvent(BeforeDial,ActionID:${INSP_ActionID}
${UNIQUEID} ${CHANNEL} ${INSP_DialInfo} ${INSP_$
exten =
dialnumber,n,Dial(${INSP_DialInfo},${INSP_RingTimeout},ghM(INSP_private^${IN
SP_ActionID}^${INSP_DialInfo}))
exten = dialnumber,n,UserEvent(AfterDial,ActionID:${INSP_ActionID}
${UNIQUEID} ${CHANNEL} ${INSP_DialInfo} ${DIALST$
exten = dialnumber,n,Hangup()
[macro-INSP_private]
exten = s,1,UserEvent(SIPDial,ActionID:${ARG1} ${UNIQUEID} ${CHANNEL}
${ARG2})
http://192.168.1.2:8088/asterisk/rawman?action=Originatechannel=Local%2Fdia
lnumber%40dialthem_private%2Fnexten=scontext=detectpriority=1CallerID=19
9async=1actionID=callingJohnSmithaccount=myaccountvaluevariable=
INSP_ActionID%3DcallJohnvariable=INSP_DialInfo%3DSIP%2F121%40flowro
utevariable=INSP_RingTimeout%3D30variable=phonenumber%3D21variabl
e=file%3DFA3469AC-BCDE-E6EB-B3AA936266704744variable=alertID%3DFA3469AC-BCD
E-E6EB-B3AA936266704744variable=subscriberID%3D1234512345
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David
Backeberg
Sent: Friday, June 26, 2009 12:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls dropping
On Thu, Jun 25, 2009 at 9:55 PM, John Regaljre...@gmail.com wrote:
When using this method, it appears that the call file creates the first
part
of the call, then creates a second call with the Dial() app. Once the call
executed by the Dial() app is answered, the two calls are joined together.
What I am experiencing is that sometime the first part of the call drops
and
therefore is never joined to the second part of the call. I see errors
like
I don't quite understand what you're trying to do, but it sounds like
call two parties and join them together. Perhaps you'd prefer to use
Originate() via AMI rather than the dialplan and extension approach
you're using now?
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-- Executing [dialnum...@dialthem_private:2]
Dial(Local/dialnum...@dialthem_private-c64a;2,
SIP/1211...@flowroute,30,ghM(dialthem_private^callJohnSmith^SIP/1211...@flowroute))
in new stack
== Using SIP RTP CoS mark 5
-- Called 1211...@flowroute
-- SIP/flowroute-0821ffc8 is making progress passing it to
Local/dialnum...@dialthem_private-c64a;2
== Spawn extension (dialthem_private, dialnumber, 2) exited non-zero on
'Local/dialnum...@dialthem_private-c64a;2'
[Jun 30 19:52:01] ERROR[26664]: pbx.c:8637 device_state_cb: Received invalid
event that had no device IE
[Jun 30 19:52:01] ERROR[26664]: app_queue.c:810 device_state_cb: Received
invalid event that had no device IE
[dialthem_private]
exten = dialnumber,1,UserEvent(BeforeDial,ActionID:${INSP_ActionID}
${UNIQUEID} ${CHANNEL} ${INSP_DialInfo} ${INSP_$
exten =
dialnumber,n,Dial(${INSP_DialInfo},${INSP_RingTimeout},ghM(INSP_private^${INSP_ActionID}^${INSP_DialInfo}))
exten = dialnumber,n,UserEvent(AfterDial,ActionID:${INSP_ActionID}
${UNIQUEID} ${CHANNEL} ${INSP_DialInfo} ${DIALST$
exten = dialnumber,n,Hangup()
[macro-INSP_private]
exten = s,1,UserEvent(SIPDial,ActionID:${ARG1} ${UNIQUEID} ${CHANNEL}
${ARG2})
[asterisk-users] Welcome Message
When I login to the asterisk, I just hear the HALF of the welcome message : You are currently the instead of You are currently the only person in the conference Thats also, I hear it after 60 secs or so.. Asterisk 1.2.27 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Welcome Message
I have gotten around the issue by adding the following to the dialplan before sending to MeetMe: exten = XXX,1,Playback(/var/lib/asterisk/sounds/silence/1) It seems to be a bug in Asterisk as far as I can tell. Hope that helps! - Josh David @ULC wrote: When I login to the asterisk, I just hear the HALF of the welcome message : You are currently the instead of You are currently the only person in the conference Thats also, I hear it after 60 secs or so.. Asterisk 1.2.27 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk module trouble
On Tue, Jun 30, 2009 at 04:57:29PM +0400, M C wrote: Hello, i have just installed asterisk 1.6.0.10 on debian 5.0 like: ./configure;make menuselect; make;make install any reason to not use the deb files ? There are no erorrs, but folder /usr/lib/asterisk/modules is empty. What am i doing wrong? Where are modules? p.s. Doing the same on Slackware, i ve got all selected modules at /usr/lib/asterisk/modules. -- But the true strength of America is found in the hearts and souls of people like Travis, people who are willing to love their neighbor, just like they would like to love themselves. - George W. Bush 02/09/2004 Springfield, MO signature.asc Description: Digital signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Welcome Message
Thanks for the Reply,
I was waiting online for someone to reply : -)
Here is my Extension file : [ Where should I enter those line ? ]
exten = 8600099,1,Meetme(8600099)
exten = 8600100,1,Meetme(8600100)
exten = 8601,1,Meetme(8601)
exten = h,1,DeadAGI(agi://127.0.0.1:4577/call_log)
exten = h,2,DeadAGI(agi://
127.0.0.1:4577/VD_hangup--HVcauses--PRI-NODEBUG-${HANGUPCAUSE}-${DIALSTATUS}-${DIALEDTIME}-${ANSWEREDTIME}
))
exten = i,1,Playback(invalid)
exten = t,1,Goto(#,1)
exten = _68600XXX,1,Meetme(${EXTEN:1},mq)
exten = _78600XXX,1,Meetme(${EXTEN:1},q)
exten = _850266.,1,Wait(2)
exten = _850266.,2,Voicemail(${EXTEN:14})
exten = _850266.,3,Hangup()
exten = _851X,1,Answer()
exten = _851X,2,Playback(${EXTEN})
exten = _851X,3,Hangup()
exten = _90009.,1,Answer()
exten = _90009.,2,AGI(agi-VDADcloser.agi,${EXTEN}-START)
exten = _90009.,3,Hangup()
exten = _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _9X.,2,Dial(SIP/${EXTEN:1...@sip8||tTor)
exten = _9X.,3,Hangup()
exten = _8X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _8X.,2,Dial(SIP/${EXTEN:1...@sip209||tTor)
exten = _8X.,3,Hangup()
exten = _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten = _X38600XXX,2,Hangup()
exten = _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten = _X48600XXX,2,Hangup()
exten = _[1-7]X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _[1-7]X.,2,Dial(SIP/${ext...@sip8||tTor)
exten = _[1-7]X.,3,Hangup()
On Wed, Jul 1, 2009 at 6:19 AM, David @ULC ucoms2...@gmail.com wrote:
When I login to the asterisk, I just hear the HALF of the welcome message :
You are currently the instead of You are currently the only person in
the conference
Thats also, I hear it after 60 secs or so..
Asterisk 1.2.27
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Re: [asterisk-users] Echo and static on PRI with errors
What do you do if you find things sharing interrupts (IRQ 11) in my case with my X100P card. I believe there is some sort of internal audio card in my cheap slow PC. Alejandro Kauffmann wrote: Tom O'Connor wrote: On Tue, Jun 30, 2009 at 3:12 PM, Tilghman Lesher tilgh...@mail.jeffandtilghman.com mailto:tilgh...@mail.jeffandtilghman.com wrote: On Tuesday 30 June 2009 08:24:29 Tom O'Connor wrote: I'm currently pointing fingers at either the hardware (someone on #asterisk said it could be a cruddy chipset, but it's an HP Server.. so should be kosher.. ), I Is it an HP server from the HP server line, or is it an HP server from the old Compaq line? Don't assume that because of the HP name, it's actually reliable with 3rd party hardware. It's a HP DL145 G2. more than that, i can't say. -- Tom O'Connor http://www.twinhelix.org t...@twinhelix.org mailto:t...@twinhelix.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The card is TE110P compatible and as such probably suffers from the same interrupt sharing problem. The ...HDLC Bad FCS.. messages tend to be related to interrupt sharing. What does lspci -vb show? Anything sharing interrupts with the card? Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- John F. Ervin *Central Florida TeleSource, LLC.** *4270 Aloma Ave #124-69C Winter Park, FL 32792 (W) 407-679-6238 (F) 866-566-1282 (F) 321-445-0781 jer...@jervin.com mailto:jer...@jervin.com http://jervin.com/cft ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
