[asterisk-users] OT - mstts.agi - Where to find API key ?
Hi, I recently discovered http://zaf.github.com/asterisk-mstts/ . In the page above, it is mentioned you have to subscribe to Microsoft Translator API on Azure Marketplace. In Azure Marketplace, I found something called Microsoft Translator. This API is free within a 2 000 000 characters per onth limit. Is this the API needed for MS TTS ? If not, where and how can I find the good one ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using disallowed_methods in sip.conf
Hi, I don't want to use INFO for DTMF, I want to transmit it inband. Therefore I set dtmfmode=inband in the sip.conf general section. But that didn't make it. As my UAC offered INFO in the Allowed-Header and Asterisk offered INFO in the 200 OK, DTMF is transmitted via INFO. Additionally I then set disallowed_methods=INFO in the sip.conf general section, but Asterisk is still offering INFO in the Allowed-Header. Has someone experience the same? Am I doing something wrong or is it a bug that INFO is still offered although it should be disallowed. If someone has managed to use the disallowed_methods options successfully, please tell me your configuration and your Asterisk Version. I am using Asterisk 10.5.0. Greetings, Paul -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another IP address to block
Where can I find such ip-lists, please? Am 05.06.2012 18:40, schrieb Alejandro Imass: We use complete regional blocks from Wizcraft and blocking at minimum all of unwanted Asia Pacific, Nigeria, Middle East, Russia, etc. We block almost anything that is not our actual customer market and screw the rest. On Tue, Jun 5, 2012 at 12:14 PM, Carlos Chavezcur...@telecomabmex.com wrote: Yesterday a customer was attacked from the following IP addresses so add them to your blacklist: iptables -A INPUT -s 37.8.119.75 -j DROP iptables -A INPUT -s 37.8.22.240 -j DROP -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Another IP address to block
On 06-06-12 11:41, Thorsten Göllner wrote: Where can I find such ip-lists, please? http://www.ipdeny.com/ Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with OpenSMS API?
Hello Thorsten Göllner, Am 2012-06-04 11:16:58, hacktest Du folgendes herunter: What do you want to do? Sending and receiving SMS? Yes The OpenSMS Server is located on the Vodafone EasyBox 803 A which is connected trough the ISDN Cable to my ISDN Card and via the Ethernet to my Switch Now I do not know, how to get the SMS from the Huawei K3765-HV USB-Stick which is on the EasyBox. I have found infos on the Net, that asterisk can do it, but on the other side I have gotten infos about http protocol... And there is more then one implementation of the OpenSMS API (at least three different types) However, connecting the USB-Sticks over a TT-Hub (Tetra-Hub) does not very well, because if ONE USB-Stick hangs, the whole machine has to reboot. This is, why I have choosen the way using some cheap EasyBoxes, and also allow me in case of trouble to connect an analog telephone. Thanks, Greetings and nice Day/Evening Michelle Konzack -- # Debian GNU/Linux Consultant ## Development of Intranet and Embedded Systems with Debian GNU/Linux Internet Service Provider, Cloud Computing http://www.itsystems.tamay-dogan.net/ itsystems@tdnet Jabber linux4miche...@jabber.ccc.de Owner Michelle Konzack Gewerbe Strasse 3 Tel office: +49-176-86004575 77694 Kehl Tel mobil: +49-177-9351947 Germany Tel mobil: +33-6-61925193 (France) USt-ID: DE 278 049 239 Linux-User #280138 with the Linux Counter, http://counter.li.org/ signature.pgp Description: Digital signature -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - mstts.agi - Where to find API key ?
I stricly followed instructions steps 1 and 2 and I'm very to report it works ! Thanks for your detailed answer. May I post here suggestions that may help others to use this script ? 2012/6/6, Lefteris Zafiris zaf@gmail.com: On 06/06/2012 10:47 AM, Olivier wrote: Hi, I recently discovered http://zaf.github.com/asterisk-mstts/ . In the page above, it is mentioned you have to subscribe to Microsoft Translator API on Azure Marketplace. In Azure Marketplace, I found something called Microsoft Translator. This API is free within a 2 000 000 characters per onth limit. Is this the API needed for MS TTS ? If not, where and how can I find the good one ? Regards The steps required to get the credentials are described here: http://msdn.microsoft.com/en-us/library/hh454950.aspx (steps 1 and 2). Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card Issue SOLVED
For some reason 1.4.4.x was not reading chan_dahdi.conf. When I symlinked it to zapata.conf it worked. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, May 30, 2012 2:35 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sangoma Card Issue Has anyone experienced an issue with Sangoma analog cards where the card suddenly stops working? Trying to dial out shows the channel as busy, even though there is no active call on that port? This happened to us often when we used Digium cards (in fact this issue is why we stopped using Digium). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Card Issue SOLVED
On 06/06/2012 09:46 AM, Eric Wieling wrote: For some reason 1.4.4.x was not reading chan_dahdi.conf. When I symlinked it to zapata.conf it worked. That means Asterisk (1.4.4.x?) was built against Zaptel, not DAHDI. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Session-timers and TCP
We have attempted to upgrade to the latest 1.8 release as well as the latest asterisk 10 release and are experiencing the same issue with session-timers and TCP where the session-timers seem to trigger and go out according to the CLI, but we are not seeing that traffic via tcpdump on the asterisk server and the call stays up when it should be disconnected. Anyone have any ideas? If more information is needed I would be happy to provide, just let me know what may help. Thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aaron Hamstra Sent: Friday, June 01, 2012 1:28 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Session-timers and TCP All, We are having issues with one of our customers. They typically are using remote sip clients on smart phones. For the purpose of allowing the apps to work properly in the background we have to utilize TCP which works fine. The problem comes up when the softphone loses connectivity for some reason. The session timers are not ending the call as they do on a UDP session. Basically from the sip debug it sends the re-invite for the session timer according to the sip debug and it appears all is fine instead of not getting a response back from the client and disconnecting the call as it does with udp. There is no way it is getting a response back from the client however as the client has no network connectivity. I have run some tcpdump's on the server and when tracing the call I actually never see those re-invites going out at all from the server. We are running asterisk 1.8.7.0 currently. I can reproduce the issue at will by establishing a call from a softphone and then putting it into airplane mode to simulate the connectivity loss. Are session-timers expected to work with tcp? If so can anyone tell me where to look to see what might be going on? Thanks in Advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OOh323 log fills with : In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
Michelle,
I forwarded your message to the OOH323 maintainer / developer. Here is
his reply.
Vladimir, there is 1.6 asterisk which is unsupported already, you can
recommend upgrade to 1.8 or higher version. I can produce patch for 1.6
but upgrade is better way
Thank you,
Vladimir
On 6/5/2012 8:58 AM, Michelle Dupuis wrote:
We have an Ast 1.6 installation which is connected to an Avaya using
ooh323. Something is causing the log to fill with In ooEndCall call
state is - OO_CALL_CLEAR (incoming, ooh323c_1) messages every 100ms.
This causes the log to grow to 300MB in just 5 minutes, which
eventually overloads the box.
Looking through the ooh323 log below, I suspect this stems from the
Error:Failed to enqueue ReleaseComplete message to outbound
queue.(incoming, ooh323c_1) message - but we don't don't see enough
H323 installations to dig deeper. Can someone offer some suggested
causes and resolutions?
Thanks!
-Date 06/01/12-
02:45:33:447 New connection at H225 receiver
02:45:33:447 Created a new call (incoming, ooh323c_1)
02:45:33:463 Receiving H.2250 message (incoming, ooh323c_1)
02:45:33:463 H.2250 message length is 12
02:45:33:463 Received Q.931 message: (incoming, ooh323c_1)
02:45:33:463 Received H.2250 Message = {
02:45:33:463 protocolDiscriminator = 8
02:45:33:463 callReference = 0
02:45:33:463 from = originator
02:45:33:463 messageType = 7d
02:45:33:464 Cause IE = {
02:45:33:464Q931NormalUnspecified
02:45:33:464 }
02:45:33:464 No UserUser IE found in ooDecodeUUIE
02:45:33:464 Error:Failed to decode received H.2250 message.
(incoming, ooh323c_1)
02:45:33:464 Decoded Q931 message (incoming, ooh323c_1)
02:45:33:464 }
02:45:33:464 ERROR:Failed ooH2250Receive - Clearing call (incoming,
ooh323c_1)
02:45:33:464 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:464 Building Release Complete message to send(incoming,
ooh323c_1)
02:45:33:464 Built Release Complete message (incoming, ooh323c_1)
02:45:33:479 Asn1Error: -4 at ooh323c/src/encode.c:584
02:45:33:479 ERROR: UserInfo encoding failed
02:45:33:479 Error:Failed to encode uuie. (incoming, ooh323c_1)
02:45:33:479 Error:Failed to encode H225 message. (incoming, ooh323c_1)
02:45:33:479 Error:Failed to enqueue ReleaseComplete message to
outbound queue.(incoming, ooh323c_1)
02:45:33:479 Receiving H.2250 message (incoming, ooh323c_1)
02:45:33:479 H.2250 message length is 12
02:45:33:479 Received Q.931 message: (incoming, ooh323c_1)
02:45:33:479 Received H.2250 Message = {
02:45:33:479 protocolDiscriminator = 8
02:45:33:479 callReference = 0
02:45:33:479 from = originator
02:45:33:479 messageType = 7d
02:45:33:479 Cause IE = {
02:45:33:479Q931NormalCallClearing
02:45:33:479 }
02:45:33:479 No UserUser IE found in ooDecodeUUIE
02:45:33:479 Error:Failed to decode received H.2250 message.
(incoming, ooh323c_1)
02:45:33:479 Decoded Q931 message (incoming, ooh323c_1)
02:45:33:479 }
02:45:33:479 ERROR:Failed ooH2250Receive - Clearing call (incoming,
ooh323c_1)
02:45:33:479 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:579 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:679 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:779 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:879 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:979 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:079 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:179 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:279 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:379 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:479 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:579 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:679 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:779 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:879 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:979 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:35:079 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:35:179 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:35:279 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:35:379 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:35:479 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:35:579 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:35:679 In ooEndCall call state is -
Re: [asterisk-users] OOh323 log fills with : In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
Vladimir:
Thanks for that! Does the response mean that the fix is already in the latest
Asterisk 1.8 ditribution?
I would prefer not to upgrade this site to 1.8 (since it requires retesting
lots of customer code)...so a patch would be ideal.
Thanks,
Michelle
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir Mikhelson
[v...@mikhelson.com]
Sent: Wednesday, June 06, 2012 11:18 AM
To: Asterisk Users List
Subject: Re: [asterisk-users] OOh323 log fills with : In ooEndCall call state
is - OO_CALL_CLEAR (incoming, ooh323c_1)
Michelle,
I forwarded your message to the OOH323 maintainer / developer. Here is his
reply.
Vladimir, there is 1.6 asterisk which is unsupported already, you can recommend
upgrade to 1.8 or higher version. I can produce patch for 1.6 but upgrade is
better way
Thank you,
Vladimir
On 6/5/2012 8:58 AM, Michelle Dupuis wrote:
We have an Ast 1.6 installation which is connected to an Avaya using ooh323.
Something is causing the log to fill with In ooEndCall call state is -
OO_CALL_CLEAR (incoming, ooh323c_1) messages every 100ms. This causes the log
to grow to 300MB in just 5 minutes, which eventually overloads the box.
Looking through the ooh323 log below, I suspect this stems from the
Error:Failed to enqueue ReleaseComplete message to outbound queue.(incoming,
ooh323c_1) message - but we don't don't see enough H323 installations to dig
deeper. Can someone offer some suggested causes and resolutions?
Thanks!
-Date 06/01/12-
02:45:33:447 New connection at H225 receiver
02:45:33:447 Created a new call (incoming, ooh323c_1)
02:45:33:463 Receiving H.2250 message (incoming, ooh323c_1)
02:45:33:463 H.2250 message length is 12
02:45:33:463 Received Q.931 message: (incoming, ooh323c_1)
02:45:33:463 Received H.2250 Message = {
02:45:33:463 protocolDiscriminator = 8
02:45:33:463 callReference = 0
02:45:33:463 from = originator
02:45:33:463 messageType = 7d
02:45:33:464 Cause IE = {
02:45:33:464Q931NormalUnspecified
02:45:33:464 }
02:45:33:464 No UserUser IE found in ooDecodeUUIE
02:45:33:464 Error:Failed to decode received H.2250 message. (incoming,
ooh323c_1)
02:45:33:464 Decoded Q931 message (incoming, ooh323c_1)
02:45:33:464 }
02:45:33:464 ERROR:Failed ooH2250Receive - Clearing call (incoming, ooh323c_1)
02:45:33:464 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:464 Building Release Complete message to send(incoming, ooh323c_1)
02:45:33:464 Built Release Complete message (incoming, ooh323c_1)
02:45:33:479 Asn1Error: -4 at ooh323c/src/encode.c:584
02:45:33:479 ERROR: UserInfo encoding failed
02:45:33:479 Error:Failed to encode uuie. (incoming, ooh323c_1)
02:45:33:479 Error:Failed to encode H225 message. (incoming, ooh323c_1)
02:45:33:479 Error:Failed to enqueue ReleaseComplete message to outbound
queue.(incoming, ooh323c_1)
02:45:33:479 Receiving H.2250 message (incoming, ooh323c_1)
02:45:33:479 H.2250 message length is 12
02:45:33:479 Received Q.931 message: (incoming, ooh323c_1)
02:45:33:479 Received H.2250 Message = {
02:45:33:479 protocolDiscriminator = 8
02:45:33:479 callReference = 0
02:45:33:479 from = originator
02:45:33:479 messageType = 7d
02:45:33:479 Cause IE = {
02:45:33:479Q931NormalCallClearing
02:45:33:479 }
02:45:33:479 No UserUser IE found in ooDecodeUUIE
02:45:33:479 Error:Failed to decode received H.2250 message. (incoming,
ooh323c_1)
02:45:33:479 Decoded Q931 message (incoming, ooh323c_1)
02:45:33:479 }
02:45:33:479 ERROR:Failed ooH2250Receive - Clearing call (incoming, ooh323c_1)
02:45:33:479 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:579 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:679 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:779 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:879 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:33:979 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:079 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:179 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:279 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:379 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:479 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:579 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:679 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:779 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:879 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
02:45:34:979 In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
Re: [asterisk-users] CDRs on multiple servers.
Owais Ahmad millennium@gmail.com writes: Hello guys, I need to be able to throw cdrs on more than one servers at a time. Please let me know how this can be done. cdr_adaptive_odbc handles multiple servers. Just define several with [foo] and [bar] and it Just Works. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OOh323 log fills with : In ooEndCall call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
Michelle,
I have re-sent your message to the developer. I will let you know when
I get a reply.
I do believe this fix is incorporated in 1.8, and all ongoing
development is done in 1.8+
Thank you,
Vladimir
On 6/6/2012 10:26 AM, Michelle Dupuis wrote:
Vladimir:
Thanks for that! Does the response mean that the fix is already in
the latest Asterisk 1.8 ditribution?
I would prefer not to upgrade this site to 1.8 (since it requires
retesting lots of customer code)...so a patch would be ideal.
Thanks,
Michelle
*From:* asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Vladimir
Mikhelson [v...@mikhelson.com]
*Sent:* Wednesday, June 06, 2012 11:18 AM
*To:* Asterisk Users List
*Subject:* Re: [asterisk-users] OOh323 log fills with : In ooEndCall
call state is - OO_CALL_CLEAR (incoming, ooh323c_1)
Michelle,
I forwarded your message to the OOH323 maintainer / developer. Here
is his reply.
Vladimir, there is 1.6 asterisk which is unsupported already, you can
recommend upgrade to 1.8 or higher version. I can produce patch for
1.6 but upgrade is better way
Thank you,
Vladimir
On 6/5/2012 8:58 AM, Michelle Dupuis wrote:
We have an Ast 1.6 installation which is connected to an Avaya using
ooh323. Something is causing the log to fill with In ooEndCall call
state is - OO_CALL_CLEAR (incoming, ooh323c_1) messages every
100ms. This causes the log to grow to 300MB in just 5 minutes, which
eventually overloads the box.
Looking through the ooh323 log below, I suspect this stems from the
Error:Failed to enqueue ReleaseComplete message to outbound
queue.(incoming, ooh323c_1) message - but we don't don't see enough
H323 installations to dig deeper. Can someone offer some suggested
causes and resolutions?
Thanks!
-Date 06/01/12-
02:45:33:447 New connection at H225 receiver
02:45:33:447 Created a new call (incoming, ooh323c_1)
02:45:33:463 Receiving H.2250 message (incoming, ooh323c_1)
02:45:33:463 H.2250 message length is 12
02:45:33:463 Received Q.931 message: (incoming, ooh323c_1)
02:45:33:463 Received H.2250 Message = {
02:45:33:463 protocolDiscriminator = 8
02:45:33:463 callReference = 0
02:45:33:463 from = originator
02:45:33:463 messageType = 7d
02:45:33:464 Cause IE = {
02:45:33:464Q931NormalUnspecified
02:45:33:464 }
02:45:33:464 No UserUser IE found in ooDecodeUUIE
02:45:33:464 Error:Failed to decode received H.2250 message.
(incoming, ooh323c_1)
02:45:33:464 Decoded Q931 message (incoming, ooh323c_1)
02:45:33:464 }
02:45:33:464 ERROR:Failed ooH2250Receive - Clearing call (incoming,
ooh323c_1)
02:45:33:464 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:464 Building Release Complete message to send(incoming,
ooh323c_1)
02:45:33:464 Built Release Complete message (incoming, ooh323c_1)
02:45:33:479 Asn1Error: -4 at ooh323c/src/encode.c:584
02:45:33:479 ERROR: UserInfo encoding failed
02:45:33:479 Error:Failed to encode uuie. (incoming, ooh323c_1)
02:45:33:479 Error:Failed to encode H225 message. (incoming, ooh323c_1)
02:45:33:479 Error:Failed to enqueue ReleaseComplete message to
outbound queue.(incoming, ooh323c_1)
02:45:33:479 Receiving H.2250 message (incoming, ooh323c_1)
02:45:33:479 H.2250 message length is 12
02:45:33:479 Received Q.931 message: (incoming, ooh323c_1)
02:45:33:479 Received H.2250 Message = {
02:45:33:479 protocolDiscriminator = 8
02:45:33:479 callReference = 0
02:45:33:479 from = originator
02:45:33:479 messageType = 7d
02:45:33:479 Cause IE = {
02:45:33:479Q931NormalCallClearing
02:45:33:479 }
02:45:33:479 No UserUser IE found in ooDecodeUUIE
02:45:33:479 Error:Failed to decode received H.2250 message.
(incoming, ooh323c_1)
02:45:33:479 Decoded Q931 message (incoming, ooh323c_1)
02:45:33:479 }
02:45:33:479 ERROR:Failed ooH2250Receive - Clearing call (incoming,
ooh323c_1)
02:45:33:479 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:579 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:679 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:779 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:879 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:33:979 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:079 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:179 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:279 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:379 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
ooh323c_1)
02:45:34:479 In ooEndCall call state is - OO_CALL_CLEAR (incoming,
Re: [asterisk-users] OT - mstts.agi - Where to find API key ?
On Wed, 6 Jun 2012 16:37:01 +0200 Olivier oza_4...@yahoo.fr wrote: I stricly followed instructions steps 1 and 2 and I'm very to report it works ! I m glad you got it working. Microsoft really tried it's best to make it as complicated as possible. Thanks for your detailed answer. May I post here suggestions that may help others to use this script ? That's what this list is all about. Lefteris Zafiris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOIP PBX replacement suggestions?
The boss wants to move from landline service to VOIP service as a cost-cutting measure. We have one voice line and one fax line. The telco is billing over $100 a month for the two. We're using Hylafax for faxing and a PBX for the voice line. Our existing PBX is an Intertel Axxess box with the old v5 processor. The management and voicemail computer died years ago (PSU burned up). I'm worried that it's going to die before too much longer. We have the IPRC and several IP Phone+ devices. It's my understanding that the IP Phone+ speaks only a proprietary Intertel protocol and can never be used with any non-Intertel equipment. I would like to dump the entire Intertel box and move to Asterisk instead, but my budget for this project is exactly $0. I can't afford to buy new devices. The boss is leaning toward getting digital voice service from the local cable monopoly. They want to charge us $30 a month per line to start, and we will have to sign a 3 year contract. The monopoly in question has a reputation for very poor service, but they are a monopoly so my boss sees them as the only alternative. My worry is that if we sign that contract, we are trapped with both the intertel and the cable monopoly, and if I exceed the capacity of the intertel (or it just dies) I am SOL. My questions then are as follows: 1) Is there a way I don't know about to get Asterisk to talk to either the IPRC or the IP Phone+ directly in such a way that gets calls from one to the other? 2) Are there any reputable VOIP providers that provide business service at a rate less than $30 per line per month? The boss is adamant that we need unlimited minutes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP PBX replacement suggestions?
1) Is there a way I don't know about to get Asterisk to talk to either the IPRC or the IP Phone+ directly in such a way that gets calls from one to the other? Since you've stated that your budget is absolutely zero, I'd have to say no. It also depends on how the old system connects to the telco. If via PRI or T1, you can use a dual-port Digium card and Asterisk between the telco and the old PBX. If analog, you could do the same with a multi-port analog card. Either way, you'd have to spend some money. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.3.2: Attended transfer goes to incorrect voicemail
Hi, I've seen similar questions being asked about this issue but left unanswered. A calls B. B attended-transfers the call to C using (polycom, cisco) phone's transfer button. C does not answer the call. A gets B's voicemail. However, if B blind-transferred the call to C and C did not answer the call, A would get C's voicemail, as expected. Thanks, Hai Nguyen. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP PBX replacement suggestions?
On Jun 6, 2012, at 3:06 PM, Doug Lytle wrote: 1) Is there a way I don't know about to get Asterisk to talk to either the IPRC or the IP Phone+ directly in such a way that gets calls from one to the other? Since you've stated that your budget is absolutely zero, I'd have to say no. It also depends on how the old system connects to the telco. It has analog lines now, but it used to have a T1 interface. I still have the T1 card. I run a 4-port Sangoma T1 card in our router PC, of which 3 ports are available. That card doesn't have an echo canceler though. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP PBX replacement suggestions?
--Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Seagraves Sent: Wednesday, June 06, 2012 2:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VOIP PBX replacement suggestions? The boss wants to move from landline service to VOIP service as a cost-cutting measure. We have one voice line and one fax line. The telco is billing over $100 a month for the two. We're using Hylafax for faxing and a PBX for the voice line. Our existing PBX is an Intertel Axxess box with the old v5 processor. The management and voicemail computer died years ago (PSU burned up). I'm worried that it's going to die before too much longer. We have the IPRC and several IP Phone+ devices. It's my understanding that the IP Phone+ speaks only a proprietary Intertel protocol and can never be used with any non-Intertel equipment. I would like to dump the entire Intertel box and move to Asterisk instead, but my budget for this project is exactly $0. I can't afford to buy new devices. The boss is leaning toward getting digital voice service from the local cable monopoly. They want to charge us $30 a month per line to start, and we will have to sign a 3 year contract. The monopoly in question has a reputation for very poor service, but they are a monopoly so my boss sees them as the only alternative. My worry is that if we sign that contract, we are trapped with both the intertel and the cable monopoly, and if I exceed the capacity of the intertel (or it just dies) I am SOL. My questions then are as follows: 1) Is there a way I don't know about to get Asterisk to talk to either the IPRC or the IP Phone+ directly in such a way that gets calls from one to the other? 2) Are there any reputable VOIP providers that provide business service at a rate less than $30 per line per month? The boss is adamant that we need unlimited minutes. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP PBX replacement suggestions?
--Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Seagraves Sent: Wednesday, June 06, 2012 2:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VOIP PBX replacement suggestions? The boss wants to move from landline service to VOIP service as a cost-cutting measure. We have one voice line and one fax line. The telco is billing over $100 a month for the two. We're using Hylafax for faxing and a PBX for the voice line. Our existing PBX is an Intertel Axxess box with the old v5 processor. The management and voicemail computer died years ago (PSU burned up). I'm worried that it's going to die before too much longer. We have the IPRC and several IP Phone+ devices. It's my understanding that the IP Phone+ speaks only a proprietary Intertel protocol and can never be used with any non-Intertel equipment. I would like to dump the entire Intertel box and move to Asterisk instead, but my budget for this project is exactly $0. I can't afford to buy new devices. The boss is leaning toward getting digital voice service from the local cable monopoly. They want to charge us $30 a month per line to start, and we will have to sign a 3 year contract. The monopoly in question has a reputation for very poor service, but they are a monopoly so my boss sees them as the only alternative. My worry is that if we sign that contract, we are trapped with both the intertel and the cable monopoly, and if I exceed the capacity of the intertel (or it just dies) I am SOL. My questions then are as follows: 1) Is there a way I don't know about to get Asterisk to talk to either the IPRC or the IP Phone+ directly in such a way that gets calls from one to the other? 2) Are there any reputable VOIP providers that provide business service at a rate less than $30 per line per month? The boss is adamant that we need unlimited minutes. === Where do you get your IP connection? The cable monopoly? There are several companies you can get service from. One is Teliax.com Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP PBX replacement suggestions?
Daniel Seagraves wrote: The boss wants to move from landline service to VOIP service as a cost-cutting measure. We have one voice line and one fax line. The telco is billing over $100 a month for the two. We're using Hylafax for faxing and a PBX for the voice line. Our existing PBX is an Intertel Axxess box with the old v5 processor. The management and voicemail computer died years ago (PSU burned up). I'm worried that it's going to die before too much longer. We have the IPRC and several IP Phone+ devices. It's my understanding that the IP Phone+ speaks only a proprietary Intertel protocol and can never be used with any non-Intertel equipment. I would like to dump the entire Intertel box and move to Asterisk instead, but my budget for this project is exactly $0. I can't afford to buy new devices. The boss is leaning toward getting digital voice service from the local cable monopoly. They want to charge us $30 a month per line to start, and we will have to sign a 3 year contract. The monopoly in question has a reputation for very poor service, but they are a monopoly so my boss sees them as the only alternative. My worry is that if we sign that contract, we are trapped with both the intertel and the cable monopoly, and if I exceed the capacity of the intertel (or it just dies) I am SOL. My questions then are as follows: 1) Is there a way I don't know about to get Asterisk to talk to either the IPRC or the IP Phone+ directly in such a way that gets calls from one to the other? No Intertel made sure of that long ago! 2) Are there any reputable VOIP providers that provide business service at a rate less than $30 per line per month? The boss is adamant that we need unlimited minutes. Doubtful voip.ms provides excellent service, but not unlimited minutes. it can even work into an ATA outputting an analog line, then you could go to the input of the Intertel and if/when it dies completely move to another analog system or single line phones You have been given an unreasonable charge. No budget but obtain the moon! You or your boss will live to regret getting into any contract, and with a company that already has a bad reputation even more so. One wonders how viable this business can even be, with one line, one fax, and no budget to replace an aged telephone system. I do hope you are either independently wealthy or have other prospects for employment. Just one old fart's opinion. Worth what you paid for it John Novack -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phone Inventory
Perhaps another idea (works with extensions with 3 or more digits)
#!/bin/sh
asterisk -rx sip show peers|
grep -vP '(UNKNOWN|Unmonitored)' |
cut -f1 -d/ | grep -P '\d\d\d.*' |
while read PEER
do
echo $PEER
asterisk -rx sip show peer ${PEER} |
grep -P (Useragent|Contact)
echo
done
Best regards.
--
Ing CIP. Alejandro Celi Mariátegui
a...@linux.org.pe
http://cipher.pe/web/asterisk.html
El jue, 23-02-2012 a las 09:20 -0600, Dale Noll escribió:
On 02/23/2012 08:49 AM, Danny Nicholas wrote:
Here is a snippet that somebody smarter than I am can improve upon
for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx sip
show peer $a;done|grep Useragent
for a in `asterisk -rx sip show peers|cut -f1 -d/` ;do asterisk -rx sip
show peer $a;done|grep Contact
Thanks for the inspiration!!
Here is my version, done with a single loop and gets Useragent and
Contact together with a visual separation between peers.
asterisk -rx sip show peers|
cut -f1 -d/ | grep -P '\d\d\d\d' |
grep -vP '(UNKNOWN|Unmonitored)' |
while read PEER
do
asterisk -rx sip show peer ${PEER} |
grep -P (Useragent|Contact)
echo
done
I hope others find it useful.
Dale
PS. I by no means claim to be smarter than thou. I just happen to
really like grep and the -P option ;-)
--
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Re: [asterisk-users] VOIP PBX replacement suggestions?
On Wed, Jun 6, 2012 at 3:59 PM, John Novack jnov...@stromberg-carlson.orgwrote: You have been given an unreasonable charge. No budget but obtain the moon! You or your boss will live to regret getting into any contract, and with a company that already has a bad reputation even more so. One wonders how viable this business can even be, with one line, one fax, and no budget to replace an aged telephone system. I do hope you are either independently wealthy or have other prospects for employment. Just one old fart's opinion. Worth what you paid for it Indeed, the amount of effort wasted in trying to save $40/mo just on this list has already exceeded that savings for many years. A business with one line and putting great effort into saving a pittance really needs to rethink its priorities. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP PBX replacement suggestions?
On 06/06/2012 12:40 PM, Daniel Seagraves wrote: The boss wants to move from landline service to VOIP service as a cost-cutting measure. We have one voice line and one fax line. The telco is billing over $100 a month for the two. We're using Hylafax for faxing and a PBX for the voice line. Unless you're going to move to an internet fax service provider you'll probably not want to attempt to switch your fax line to a VoIP line and still attempt to fax over it. And even then, depending on how much fax traffic you have moving to an internet fax service provider may not save you any money. my budget for this project is exactly $0. I can't afford to buy new devices. Unless your boss wants you to do VoIP from a headset on the PC I think you're chasing a lost cause. And, for what it's worth, $100 per month for two analog PSTN lines is rather typical. Depending on how much voice traffic you have and how much of it is local or inbound... switching to a VoIP service may not actually be a cost-cutting measure. Thanks, Lee. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
